- sr-users - kamailio.org

is kamilio support RFC5626 now ?
by mike 16 Oct '12

16 Oct '12

dear kamilio developers : i read the kamilio 3.3.0' s new features , it say it support outbound , and i have place to handle reg-id and sip.instance . so , i wonder if the kamilio3.3.0 support RFC5626 now，so that the kamilio can be used as a edge proxy . i ask why because i can't find where kamilio add flow-token and path for the register request. thanks

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IPv6 addresses in dispatcher on IPv4 only proxy
by Alex Hermann 16 Oct '12

16 Oct '12

Hello, I am using the dispatcher module with a database table shared among multiple proxies. Some proxies do both IPv4 and IPv6, others only do IPv4. The problem is when i use an IPv6 address in the dispatcher table. Then the IPv4-only proxies fail to load the table. ERROR: dispatcher [dispatch.c:325]: could not resolve [IPv6 address] WARNING: <core> [mem/f_malloc.c:474]: WARNING:fm_free: free(0) called ERROR: dispatcher [dispatcher.c:321]: could not initiate a connect to the database ERROR: <core> [sr_module.c:889]: init_mod(): Error while initializing module dispatcher (/usr/lib/kamailio/modules_k/dispatcher.so) Is there any setting i can use on the IPv4-only proxies to make it skip the IPv6 records? (apart from listening on an IPv6 address) -- Greetings, Alex Hermann

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Error while running TLS enabled Kamailio proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Dear All While executing TLS enabled Kamailio proxy (3.1.5), getting below error. * 0(15895) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(15895) ERROR: tls [tls_domain.c:393]: TLSs<default>: Unable to load certificate file './modules/tls/ser-selfsigned.pem' 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:02001002:system library:fopen:No such file or directory 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:20074002:BIO routines:FILE_CTRL:system lib 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:140DC002:SSL routines:SSL_CTX_use_certificate_chain_file:system lib 0(15895) ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module tls (/usr/local/lib/kamailio/modules/tls.so) 0(15895) ERROR: <core> [main.c:1532]: ERROR: main: error in init_child(PROC_INT) -- exiting 0(15895) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(15893) ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe * I performed below steps to compile and install. 1. I had a working Kamailio-3.1.5 code base without TLS 2. Downloaded latest openssl source code 3. Compiled and generated libssl.a and libcrypto.a 4. Next, I recompiled kamailio proxy, compilation did not go through. I had to modify Makefile in modules/tls module, added path for openssl/bio.h and path for libssl.a 5. Then executed make all clean and make all include_modules=tls , this time compilation succeeded 6. Next I did make install 7. Generated certificates using openssl library that comes by default with ubuntu 10.04 lte 8. Added below entries in kamailio.cfg #!define WITH_TLS #!ifdef WITH_TLS loadmodule "tls.so" modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/etc/certs/localb2bua.com/key.pem") modparam("tls", "certificate", "/etc/certs/localb2bua.com/cert.pem") modparam("tls", "ca_list", "/etc/certs/demoCA/cert.pem") enable_tls=yes #!endif 9. Then I tried running kamailio proxy It complained not getting tls.so and tls.cfg Copied these files from modules/tls to /usr/local/etc/kamailio/ path 10. Then I was able to run Kamailio proxy and got the errors pasted at top of email (* Unable to load certificate file './modules/tls/ser-selfsigned.pem') * Please guide me what should I do to avoid this error. Thanks a lot. -Kamal * *

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unsubscribe
by Pablo R. Digonzelli 16 Oct '12

16 Oct '12

Ing. Pablo Digonzelli Sofware Solutions IP Soluciones SRL 25 de Mayo 521. Entrepiso A email: pdigonzelli(a)softsargentina.com email: pdigonzelli(a)gmail.com twitter: @pdigonzelli Tel: 0381 4227378 Cel: 0381 155982714

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Re: [SR-Users] Kamailio 3.3.0 CDR logging call "duration"
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, do not reply to private email only when it is about a discussion on the mailing list - such messages will be ignored. Try to store the value in an avp when processing the BYE, after loose_route(), and store that avp in acc. Daniel On 10/15/12 1:13 PM, Nord7 wrote: > Hello, > no I'm not getting error messages. > Problem: > I can not get kamailio record duration dialogue created field "duration". > I make a test call, answer it, wait 5-6 seconds and hang up. > Now I have to get 5-6 seconds duration in the "duration", but the > field is blank. The rest of the field information about the call is > entered correctly. > So the variable $ DLG_lifetime information about the duration of the > call does not get. > But most of all I am wrong somewhere. > > > 2012/10/15 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > do you get any error message or what is the problem you encounter? > > Cheers, > Daniel > > > On 10/15/12 10:14 AM, Nord7 wrote: >> Hello >> >> Kamailio 3.3.0 >> All default "extra fields" created in database. I create in >> Database new field "duration" and trying write in this field >> variable $DLG_lifetime (from dialog module) , but can't do this. >> What i'm doing wrong? >> >> installed modules: ACC, DIALOG >> # ----- acc params ----- >> /* what special events should be accounted ? */ >> modparam("acc", "early_media", 0) >> modparam("acc", "report_ack", 0) >> modparam("acc", "report_cancels", 0) >> modparam("acc", "detect_direction", 0) >> /* account triggers (flags) */ >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", FLT_ACC) >> modparam("acc", "log_missed_flag", FLT_ACCMISSED) >> modparam("acc", "log_extra", >> "src_user=$fU;src_domain=$fd;src_ip=$si;" >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) >> /* enhanced DB accounting */ >> 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;src_ip=$si;" /* extra fields >> created in DataBase */ >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "cdr_enable", 1) >> modparam("acc", "cdr_start_on_confirmed", 0) >> modparam("acc", "cdr_facility", "LOG_DAEMON") >> >> # ----- dialog params ----- >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "hash_size", 4096) >> modparam("dialog", "rr_param", "did") >> modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ >> modparam("dialog", "default_timeout", 21600) >> modparam("dialog", "timeout_avp", "$avp(i:10)") >> modparam("dialog", "dlg_match_mode", 0) >> modparam("dialog", "detect_spirals", 1) >> modparam("dialog", "db_url", DBURL) >> modparam("dialog", "db_mode", 1) >> modparam("dialog", "db_fetch_rows", 500) >> >> >> ########Routing Logic######## >> >> route[RELAY] { >> >> # enable additional event routes for forwarded requests >> # - serial forking, RTP relaying handling, a.s.o. >> if (is_method("INVITE|SUBSCRIBE")) { >> t_on_branch("MANAGE_BRANCH"); >> t_on_reply("MANAGE_REPLY"); >> } >> if (is_method("INVITE")) { >> t_on_failure("MANAGE_FAILURE"); >> } >> setflag(DLG_FLAG); # flag 4 duration >> setflag(FLT_ACC); # do accounting ... >> setflag(FLT_ACCFAILED); # ... even if the transaction fails >> 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 > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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segfault in siptrace module using setflag?
by samuel 16 Oct '12

16 Oct '12

Hi folks, I was making some tests with kamailio (both 3.3.0 and 3.3.1 tgz) with siptrace module sending "homer" packets to a remote central agent where the data is stored and presented to the user. The related config is: # ----- siptrace params from HOMER ----- modparam("siptrace", "duplicate_uri", "sip:A.B.C.D:9060") modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #HEPv2 == timestamp will be included to HEP header modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_capture_id", 1) basically for all incoming messages, a route is called with the following actions: route[HOMER] { setflag(22); sip_trace(); } I've been making some tests and if I disable the setflag action, there is no problem (well, no transaction messages are sent to the central agent), but as soon as I activated setflag, kamailio crashes with the following error: <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 28 There's no concrete point in the config file where it fails so I think the problem is more tm-internals related. I can provide further information in case you need it. Just for adding information, and without any intention to start a flame-war, with opensips 1.8.0-tls the server does not crash... Thanks in advance! Samuel.

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Supporting TLS and DTLS in RTP Proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Hi All I am planning to enhance RTP proxy to support TLS and DTLS. We have some requirements where we need to send RTP packets either over TLS or over DTLS. I really do not know where to start :-) Few things that I have thought over is 1. I will take opnssl code base and integrate with rtpproxy code. 2. Will refer kamailio proxy code how tls connection framework is there and try to copy the code from kamailio code base to rtp proxy code base, OR is there a better alternative here. If somebody has a better idea, kindly share. Thanks a lot. -kamal

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Guidance on using Kamailio as Presence Server and Scalability issues for millions of subsribers
by Thomas Hoffman 16 Oct '12

16 Oct '12

We're trying to determine how best to design/architect a solution where we use Kamailio as a Presence/XCAP server and be scalable to support millions of subscribers. I see the Dispatcher module, but it appears that requires having dedicated instances of the server running just in Dispatcher mode - then you can load balance across the actual instances of the presence servers. It's not clear if a server can be configured to work as a dispatcher and a presence server at the same time. Does anyone have any high-level guidance how to approach this problem? Thank you, Tom Hoffman ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

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ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1)
by MARINA SERRANO MONTES 16 Oct '12

16 Oct '12

Hi, My kamailio installation usually works right, but sometimes, it crashes and it shows the following error: "ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) " Do you know why it happens randomly or what's the cause ? Many thanks. Br, Marina ________________________________ 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 enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

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preparing kamailio 3.3.2
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, I am considering packaging Kamailio v3.3.2 at the beginning of next week (Monday or Tuesday). If any of the devs have backports to be done, do them before the end of this week. Also, if someone spotted some fixes in the master that were not backported, write to the lists so we can care of them. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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Delay message
by pars3c 15 Oct '12

15 Oct '12

Hi, how can i delay sending of a message ? A ----> proxy (after x seconds) ---> B If I use the sleep function of the cfgutils module,I block all the calls for the sleep time or the single one? Thanks

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Error on compiling Kamailio with TLS support in Ubuntu 10.04LTE
by Kamal Palei 15 Oct '12

15 Oct '12

Hi All I am trying to compile Kamailio - 3.1.5 in Ubuntu 10.04LTE . Tried below comands one by one make -C modules/tls make modules modules=modules/tls make all include_modules=tls But always getting the error *In file included from tls_bio.c:31: tls_bio.h:32:25: warning: openssl/bio.h: No such file or directory * It looks that openssl is not installed in machine. I tried installing openssl as belowB sudo apt-get install libssl-dev But it fails always. The error it gets is *root@B2BUA:/usr/local/src/kamailio-3.1.5# sudo apt-get install libssl-dev sudo: unable to resolve host B2BUA Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: libssl-dev: Depends: libssl0.9.8 (= 0.9.8k-7ubuntu8) but 0.9.8k-7ubuntu8.8 is to be installed E: Broken packages* * root@B2BUA:/usr/local/src/kamailio-3.1.5# * Has anybody faced this issue previously. Whats the workaround here, if somebody knows please share. Thanks Kamal

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Re: [SR-Users] R: Delay message
by Alex Balashov 15 Oct '12

15 Oct '12

You can't, exactly. But you can t_continue() inside an rtimer route that fires at a given interval. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/pars3c <pars3c(a)gmail.com> wrote:With t_suspend of tmx module, how can I set seconds to sleep? -----Messaggio originale----- Da: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Per conto di Alex Balashov Inviato: lunedì 15 ottobre 2012 15:52 A: sr-users(a)lists.sip-router.org Oggetto: Re: [SR-Users] Delay message It sounds like you're after the recent asynchronous TM innovations: http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2535968 http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2525882 On 10/15/2012 09:49 AM, pars3c wrote: > Hi, > > how can i delay sending of a message ? > > A ----> proxy (after x seconds) ---> B > > If I use the sleep function of the cfgutils module,I block all the > calls for the sleep time or the single one? > > Thanks > > > > _______________________________________________ > 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 > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ----- Nessun virus nel messaggio. Controllato da AVG - www.avg.com Versione: 2012.0.2221 / Database dei virus: 2441/5332 - Data di rilascio: 14/10/2012 _______________________________________________ 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

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Kamailio 3.3.0 CDR logging call "duration"
by Nord7 15 Oct '12

15 Oct '12

Hello Kamailio 3.3.0 All default "extra fields" created in database. I create in Database new field "duration" and trying write in this field variable $DLG_lifetime (from dialog module) , but can't do this. What i'm doing wrong? installed modules: ACC, DIALOG # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_level", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ 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;src_ip=$si;" /* extra fields created in DataBase */ "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_DAEMON") # ----- dialog params ----- modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ modparam("dialog", "default_timeout", 21600) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "db_fetch_rows", 500) ########Routing Logic######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } setflag(DLG_FLAG); # flag 4 duration setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails if (!t_relay()) { sl_reply_error(); } exit; }

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DTLS support in Kamailio
by Kamal Palei 15 Oct '12

15 Oct '12

Hi List Is DTLS supported in Kamailio? If not, can somebody give me initial idea/pointer how can we support it with minimal code change. Best Regards Kamal

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Re: [SR-Users] Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 15 Oct '12

15 Oct '12

Hello I have created a SRV record for sip.domain.tld pointing to our kamailio server, the domain by itself resolves to our DNS IP. Regarding the sockets: root@kamailio1:~# netstat -lnp | grep kamailio tcp 0 0 192.168.14.25:5060 0.0.0.0:* LISTEN 4752/kamailio unix 2 [ ACC ] STREAM LISTENING 17100 4745/kamailio /tmp/kamailio_ctl Thanks 2012/10/11 Daniel-Constantin Mierla <miconda(a)gmail.com> > Hello, > > I recommend using 3.3.x if you start a new deployment. > > Is the domain resolvable by DNS? What are the sockets kamailio is > listening on? > > Cheers, > Daniel > > On 10/10/12 5:21 PM, Christophe ROY wrote: > > Hello > > I'm quite new to SIP and TOIP so I may not understand the concepts > correctly. > I'm supposed to make work SIP clients (for now, linphone on android and > windows) with a Kamailio 3.2 server and ldap authentication. > > Ldap authentication seems to be working fine, as the login/pass is > validated against the Active Directory before registering the SIP client. > However, Linphone seems to register the SIP client as > sip:user@true_ip_address instead of sip:user@domain. > > If I try to make a call to sip:anotheruser@domain it doesn't work, and I > can find in /var/log/syslog the following lines: > > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: > no corresponding socket for af 2 > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding > listening socket) > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl > [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server > error occurred (7/SL) > > However, if I try to make a direct call to sip:anotheruser@his_true_IP, > it works > > How can I make the registered users to be reachable with user@domain with > Kamailio? > And by the way, is there a way to get a list of registered users? > > Thank you for the help > > Christophe > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu > > > _______________________________________________ > 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 > >

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Wesip Netbeans
by pars3c 15 Oct '12

15 Oct '12

Hi , anyone know if it is possible to use Wesip with Glassfish instead of Tomcat? I'm new in using Glassfish and sip servlet, and I don't know how to start the sip servlet. In tomcat there is the file server.xml, and the connector . How "say" to glassfish to start a listener on sip packet ? If it's not possible to use Glassfish, how i can open a sar project with netbeans e debug it? Thanks to all

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update_subs_rlsdb: NO MATCH
by Sangeeta Shah 12 Oct '12

12 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. I see this information message in the log file: Oct 8 14:34:05 RCS-Presence /usr/local/sbin/kamailio[4530]: INFO: rls [rls_db.c:779]: update_subs_rlsdb: NO MATCH Anyone know why the application is calling that method if it doesn't exist in rls_db.c. I still haven't been able to get presence with rls and integrated xcap working so not sure if this is an issue or no. Currently for both presence and RLS i am using "DB-Only" mode. Any help is appreciated. It seems like when client A adds client B as a contact, the kamalio server is not sending out an auth request to client B. Even though it's registered for presence.winfo events. Thanks, Sangeeta -- Sangeeta Shah

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TLS support in Kamailio
by Kamal Palei 12 Oct '12

12 Oct '12

Hi List I have a specific requiremnt where I want Kamailio proxy need to work as a intermideiate proxy . The endpoints will talk to Kamailio using UDP as transport and Kamailio should interact with main proxy using TLS. Is this scenario possible with Kamailio ? If so, can you give me some pointer how can I acheive it. Thanks Kamal Net clooud systems Bangalore

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Kamailio Group voice Call
by Vijay Thakur 12 Oct '12

12 Oct '12

All Experts, I have setup kamailio 3.1.5 server and it is working fine in my production environment. I want to implement group voice calls (audio conferencing). How can i do this. Kindly guide me. With Regards, Vijay Thakur

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Skype-like service in less than one hour
by శ్రీధర్ 12 Oct '12

12 Oct '12

I set up my server following the tutorial at http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, with the latest kamailio 3.3, and updated the config file accordingly. After the reboot the sip server is not running here are the logs. When I do 'sudo /etc/init.d/kamailio start' the server starts, but has problems communicating with the rtpproxy, the log for the same is also below. Please advice if any changes needed to make this working with the latest version Thanks S $ cat /var/log/syslog | grep kamailio Oct 12 01:07:48 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:07:48 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [db.c:320]: could not add connection to the pool Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: usrloc [dlist.c:513]: failed to open database connection Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: registrar [reg_mod.c:472]: failed to register domain Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:623 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:626 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:628 Oct 12 01:54:55 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:54:55 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSs<default>: tls_method=12 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSs<default>: certificate='/etc/kamailio/kamailio-selfsigned.pem' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSs<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSs<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSs<default>: require_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSs<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSs<default>: private_key='/etc/kamailio/kamailio-selfsigned.key' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSs<default>: verify_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSs<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:576]: TLSs<default>: No client certificate required and no checks performed Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSc<default>: tls_method=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSc<default>: certificate='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSc<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSc<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSc<default>: require_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSc<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSc<default>: private_key='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSc<default>: verify_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSc<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:561]: TLSc<default>: Server MUST present valid certificate Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily

4 6

log alerting time, progress time connect time
by pars3c 12 Oct '12

12 Oct '12

Hi , i need to register the time when arrive the following messages: invite, alerting, progress , connected and hangup. How can I make it ? it's is possible to store these value in the extra field in the acc module ? Thanks

2 1

TCP error
by Ed Brady 12 Oct '12

12 Oct '12

I am testing connecting an MSRP client to my Kamailio instance, and I keep getting the following error in my syslog: /usr/sbin/kamailio[20055]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) I believe this must be a misconfiguration on my part, I just don't know where to start. Any ideas what may cause a TCP read error? If this is an uncommon error, I can supply configs to help determine the cause. Thanks, Ed

3 2

Wrong onreply_route is called after serial forking
by Alex Hermann 11 Oct '12

11 Oct '12

Hello, i noticed thet in serial forking, replies to earlier branches arriving after sending out a new branch use the onreply_route of the later branch instead of the onreply_route set before sending the earlier branch.... How to reproduce: 1) set onreply_route to A 2) relay 1st branch 3) 1st branch times out, internal 408 is created 4) tm send CANCEL to 1st branch 5) in failure route, onreply_route is set to B 6) relay 2nd branch 7) 1st branch responds with 487, and goes into reply_route B instead of A I think each branch should take the reply_route which was set before it got relayed and not pick up later changes meant for other branches. How would i fix this? -- Greetings, Alex Hermann

5 7

Question about how the RLS module handle subscribe
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello All, I am trying to understand how the RLS module handles a subscribe request. My client is sending the following: Subscribe, event = reg, no Allowed header - hence it is getting handled by the presence module (entry make its to active_watchers table) Subscribe, event - presence.winfo, no Allowed header - hence it's getting handled by the presence module (entry makes it to active_watchers table) Publish - handled by presence module (makes it to presentity table) Subscribe, event = presence, Allowed = eventlist - hence it's handled by the rls module. RLS module is having trouble sending a notify per this data in the syslog. Also I don't see any entries in the watcher table or rls_watchers or rls_presentity table. Errors: Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:387]: updating in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:273]: in function agg_body_sendn_update Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fb2a8daa220 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fb2a8daa598 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fb2a8da9f90 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fb2a8da9b48 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fb2a8da9b28 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fb2a8daa608 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fb2a8daa570 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fb2a8da9410 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === What's it trying to do here? Any help is appreciated. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio presence with RLS
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello Everyone, Has anyone had any luck getting RLS working with presence. I am seeing the following errors in my log file: ct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:385]: updating in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:271]: in function agg_body_sendn_update Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fbf415c4178 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fbf415c48a0 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fbf415c38a8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fbf415c3bc8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fbf415c4880 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fbf415c3c28 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fbf415c3648 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fbf415c3940 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === I think this is my first problem. There is an error later in the log file as well, but it seems Apparently from what i understand we are trying to update the entry in the presence_shtable and we cannot find it and hence we are failing to send the full state notify. Any help is appreciated. If further information is required please let me know. I can try to print out more data in the update_presence_shtable function. I have another htread with a lot of log info but for some reason the thread went cold and I just need to keep moving on getting this to work. Anyone familiar with this please let me know what the likely cause of this issue might be. Thanks, Sangeeta -- Sangeeta Shah

1 0

How can I directly access the domain of additional branches??
by Stephen Dodge (Bistech) 11 Oct '12

11 Oct '12

Hello, Could someone advise if it is possible to directly access the domain of additional branches as $rd references the domain of the main URI only. If it's not possible does anyone have any tips on how I can achieve this. Many Thanks. Steve. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

4 5

Kamailio direct interconnectivity with PRI
by SamyGo 11 Oct '12

11 Oct '12

Hello, I've a scenario in which I've to deploy a couple Sangoma PRI cards with kamailio. What I wish is that I've some drivers for this purpose and so I don't ned to install FreeSWITCH or Asterisk in between the PRIs and Kamailio. Kindly give any feedback on what are the possibilities and options. Thanks, Sammy

4 7

Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 11 Oct '12

11 Oct '12

Hello I'm quite new to SIP and TOIP so I may not understand the concepts correctly. I'm supposed to make work SIP clients (for now, linphone on android and windows) with a Kamailio 3.2 server and ldap authentication. Ldap authentication seems to be working fine, as the login/pass is validated against the Active Directory before registering the SIP client. However, Linphone seems to register the SIP client as sip:user@true_ip_address instead of sip:user@domain. If I try to make a call to sip:anotheruser@domain it doesn't work, and I can find in /var/log/syslog the following lines: Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) However, if I try to make a direct call to sip:anotheruser@his_true_IP, it works How can I make the registered users to be reachable with user@domain with Kamailio? And by the way, is there a way to get a list of registered users? Thank you for the help Christophe

2 1

kamailio 3.3.1 - uac - remote register - init delay possible?
by René Bormann 11 Oct '12

11 Oct '12

Hello @all, want to use the remote register function of the UAC module. We need to register 42 accounts on one openSER server. At kamailio startup those 42 remote register requests will be startet at the same time. This leads to a kind of FlodAttact on the openSER server of our provider, which will block us :-( We would like to delay each initial remote register by 1 second. We searched the UAC module and taught that at some place the uac.so module has to load the uacreg table and do some for loop to start the remote register process. We found function child_init() in uac.c where the function uac_reg_load_db() is called followed uac_reg_timer(). Our change to uac_reg_timer() is the sleep(1). After compiling it seems to us - that this function is not called. Is there anybody how has a tip for us? Thanks, René /** * */ void uac_reg_timer(unsigned int ticks, unsigned int initial_reg_call) { LM_ERR("----------------------- uac_reg_timer() -- enter \n"); <------>int i; <------>reg_item_t *it = NULL; <------>time_t tn; <------>tn = time(NULL); <------>for(i=0; i<_reg_htable->htsize; i++) <------>{ <------><------>/* free entries */ <------><------>it = _reg_htable->entries[i].byuuid; <------><------>while(it) <------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 1 \n"); <------><------><------>if (initial_reg_call == 1) <------><------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 2 \n"); <------><------><------><------>sleep(1); <------><------><------>} <------><------><------>uac_reg_update(it->r, tn); <------><------><------>it = it->next; <------><------>} <------>} }

3 8

MSRP Config
by Ed Brady 10 Oct '12

10 Oct '12

Hi, I am working on an MSRP project with Kamailio. I have Kamailio installed and configured for SIP with several test accounts created, but am unable to send MSRP messages so far. I am using Blink as a test client, though the final intentions of the project will be to use iDoubs and IMSDroid for client connections. We are having these developed to our needs right now though, so I must use another client to begin testing. I have been unable to find good documentation on the initial configuration of MSRP in Kamailio. I have seen the module documentation, but am unclear how to utilize the information there to configure Kamailio properly. Is there any documentation on getting started with a basic MSRP relay setup so that two clients can chat and send files? I am not new to SIP, but this is my first project with MSRP. Thanks, Ed

4 10

OpenSER/Kamailio Support? - Forwarding SIP INVITE Packets
by Eddie Elric 10 Oct '12

10 Oct '12

Hello fellow OpenSER/Kamailio users, I hope this is the right method of contact - I was unable to find a forum on which to post any queries. In essence, I've set up a Kamailio server of the latest version - logging on to clients works fine, etc. The idea of my setting up was for forwarding SIP messages to our test server, that's where I got stuck. The following link explains everything on Kamailio it seems - http://www.kamailio.org/wiki/cookbooks/3.3.x/core#forward So, I've added to the kamailio.cfg the following - forward("10.0.0.10","5060"); But you see, I'm not sure where I should have added it. I've tried adding to a variety of places, such as - route[SIPOUT] and onreply_route[MANAGE_REPLY] - but none are working. Cheers guys/girls. Best, Eddie

1 1

Kamailio/RTP Proxy issue
by Kamal Palei 10 Oct '12

10 Oct '12

Hi All Facing a strange issue in Kamailio / RTP proxy setup. Using 1. Kamailio - 3.1.5 2. RTP proxy 1.1 I have pne sip client, lets say it is 'A'. 'A' sent INVITE to Kamailio proxy. Kamailio proxy modified the IP address present in SDP to RTP proxy IP address, and sent to main proxy. Main proxy did respond back with 200 OK to INVITE. In 200 OK SDP, Main proxy put the IP address of Media server (media server ip address is different than main proxy server ip address). After the call establishment, it is expected that RTP proxy should send media packets to media server, instead it is sending to proxy server. Pleae do let know, how to put a work around to avoid this problem. Thanks Kamal NECS, Bangalore

2 2

TCP replace alias can cause core
by Jijo 10 Oct '12

10 Oct '12

We found a problem regarding TCP connection alias in the following code at tcp_main.c: int tcpconn_finish_connect( struct tcp_connection* c, union sockaddr_union* from) { : : /* remove all the aliases except the first one and re-add them * (there shouldn't be more then the 3 default aliases at this * stage) */ for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); } c->aliases=1; As TCP_ALIAS_REPLACE flag is set for the default TCP options value, in the function _tcpconn_add_alias_unsafe() a TCP connection alias can be moved from connection A to connection B based on the TCP alias hash. In this case, the number of aliases is incremented in the connection A, and decremented from connection B. However, in the connection B the number of aliases can reach zero (no alias). And the code above can be executed for connection B setting the number of aliases to 1 unconditionally. When this case happens, the connection B keeps an invalid alias (already excluded from connection B by tcpconn_add_alias_unsafe() function called from connection A). When the connection A is released, the aliases are also released, and this memory area can be filled with different data. As connection B has references to an invalid alias it can try to access invalid areas, and can crash Kamailio. This access happens, for example, when another alias is added to connection B. To fix it we include a check before the code: if (c->aliases>0) { for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); memset(a,0xbb,sizeof(struct tcp_conn_alias)); } c->aliases=1; } Please let us know if any comments. Thanks Jijo

2 1

[snmpstats] openserSIPRegRejectedRegistrations not being updated
by Anatoly Ivanov 10 Oct '12

10 Oct '12

Hi, I am observing weird behavior of snmpstats.so - I see openserSIPRegAcceptedRegistrations being updated, but openserSIPRegRejectedRegistrations stays at zero. Could somebody please explain under what conditions this counter is being incremented? I am using auth_radius, so I suspect I simply do not reject failed registartions the 'proper' way, thus causing the counter to stay at 0. Thank you, Anatoly.

2 3

how to use fix_nated_sdp("2") with multiple media types
by MARINA SERRANO MONTES 10 Oct '12

10 Oct '12

Hi, I have a doubt about how to use "fix_nated_sdp("2")" function when we have an scenario with multiple media types (audio and video) and the param "c" in sdp for each media type has a different IP. We are using a SCOPIA 400 MCU to multi-party in a call, and it will set up different IP to send RTP depend on media type. I have tested this function, but it rewrites always all the "c" parameter with the same value. Is there any option to use the function or to rewrite "c" value for each media type independently? Many thanks. Br, Marina ________________________________ 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 enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

3 11

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

just to complete the thread, nginx does not currently support proxying of requests that use websocket transport. it is in the roadmap, but no date is set. when nginx receives request over websocket it converts 'Upgrade: Connection: Upgrade' header to 'Connection: close' header and removes Upgrade header. -- juha

1 0

moving all to kamailio namespace
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, some bits were left with the old project name OpenSER, not to disturb too much at that time and see where everything goes. Other voices expressed same idea in the past, now everything is stable and development goes smooth, so it is good to make a decision because openser is too way back and new comers could be confused. In my list is renaming from openser to kamailio next attributes: - default database name (exiting configs should not be impacted that much as most of them have database provided by module parameter) - default database access usernames and passwords, respectively openser, openserro and openserrw should become kamailio, kamailioro and kamailiorw - perl module API is packed as OpenSER - this could affect some people using it, but a search and replace should do it - snmpstats mibs - probably it is going to have some impact as well Eventually we can start doing it gradually, one or two per release, but is probably better to do all at once, so people will do one upgrade with next major release. Another thing is sercmd, it is in conflict for packaging as both ser and kamailio install it. It means that one cannot install both kamailio and ser at same time from packages. One option is to install it as kamcmd. Because these changes affect users and devels, I sent the message to both mailing lists. Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

6 7

storing and using route-set in dialog module
by Klaus Darilion 09 Oct '12

09 Oct '12

Hi! Are there any means to store the route set of a dialog (on dialog creation) and use the stored route set when forwarding in-dialog requests instead of using the requests' route set? thanks Klaus

1 0

Force to use mediaproxy/rtpproxy
by Mino Haluz 09 Oct '12

09 Oct '12

Hi, I am doing some performance tests and I cannot force SIP INVITEs to use mediaproxy within LAN, without any NAT. I'm using engage_media_proxy or rtpproxy_manage but phones are still exchanging voice data between them, not through the proxy. Is there some trick how to forcibly tell kamailio to use media/rtp proxy ? Maybe kamailio finds out that there is no nat and therefore will not use any nat traversal mechanism. Mino

1 1

get $sht array of values
by Uri Shacked 09 Oct '12

09 Oct '12

Hi, If i have two keys on a HTABLE that are the same. i undestand i need to set the key type to 1 (array). i use the $shtcn and get 3 (size is an entry as well). so far, so good. But, now i am trying to get all the values from the table to an AVP oto say it in another way - i need all values because i parse it afterwards. So, what is the syntax? using $sht() didn't work..... Uri

2 1

kamailio cores on corrupted route header
by Jijo 09 Oct '12

09 Oct '12

Hello, kamailio cores when receives a corrupted route header. For example, this was causing the core. Route: sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=& [=<sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=&%20[=> I found the problem, the pointer was not initializing to null after freeing it. Please apply this fix in the next version. Here is the diff with the original(3.2.2) and changed version. PGA:/mnt/o/kamailio-3.2.2/parser # diff -u parse_param.c.orig parse_param.c --- parse_param.c.orig 2012-10-09 09:42:58.372003500 -0300 +++ parse_param.c 2012-10-09 21:34:14.556367900 -0300 @@ -545,6 +545,7 @@ error: if (t) pkg_free(t); free_params(*_p); + *_p = 0; return -2; ok: Thanks Jijo

1 0

Kamailio XMPP support
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Currently jabberd2 is not available as directed in wiki. Can i use openfire with kamailio for xmpp? How ? Kindly point me any direction. With Warm Wishes, Vijay Thakur information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

1 0

Kamailio XMPP Module Setup
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Kindly point me any direction. With Warm Wishes, Vijay Thakur

1 0

Kamailio FAQ Week
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, I'm looking at getting richer content for the FAQ wiki page: * http://www.kamailio.org/wiki/tutorials/faq/main Probably we can get it good enough with a bit of mobilization for one week. If anyone is having some questions fitting the purpose of the FAQ, just add them to the page or send via email. If you don't have the right answer, just leave it as TBA in the wiki, via email is fine to send just the question. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

peter suggested using nginx for dispatching xcap requests and that works fine. has someone tried using nginx to dispatch also sip over websocket requests? is there some drawbacks? -- juha

2 1

Kamailio issues errors
by Sangeeta Shah 08 Oct '12

08 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. Now I am noticing the errors in the attached syslog capture. The only thing I have changes is the DB mode to DB Only. Thanks, Sangeeta -- Sangeeta Shah

1 0

implementing call-quota with dialog module
by Aft nix 08 Oct '12

08 Oct '12

Hi, I'm trying to implement a "callquota" based on destination domains. The idea is following: every destation domain has a limit of, say 50 calls. As the sip clients are registered with the "destination domains", i've extracted the domain using "$fd" because it results as same value as destination domain. Problem is, i wanted to have separate quota for each domains. But it seems, that the quota is reached "as total" not as per domain basis. Another thing is, the number of calls reported by it is also unusually large than actual calls. This is implemenated as bellow: #!ifdef WITH_CALL_LIMIT modparam("dialog","enable_stats",1) modparam("dialog","dlg_flag", DLG_FLAG) modparam("dialog","hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) #!endif #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$fd", "$var(100)"); if ($var(100) >= 50 ) { xdbg("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); if (get_profile_size("callquota","$fd", "$var(100)")) { xdbg("DEBUG: there are $var(100) total calls for $fd"); } } #!endif Thanks in advance. -- -aft

2 2

Newbie question about RLS/XCAP/batch'd NOTIFYs
by Dmytro Bogovych 06 Oct '12

06 Oct '12

Greetings. I try to make iOS voip application with presence information. iOS does not let too frequent network updates from server side when runs in background mode. So I cannot rely on plain PUBLISH / SUBSCRIBE scheme - it leads to restart of application by iOS in many cases. Please see this post for details: https://devforums.apple.com/message/565667#565667 The question. Can Kamailio provide way to reduce the number of NOTIFYs? I see it as 1) iOS application creates & uploads RLS via XCAP 2) iOS application asks server about changes in presence information once per 10 minutes or when application goes to foreground state. The best way is to receive single batch'd NOTIFY answer for all watched peers. Is it possible to implement with kamailio? Thank you :)

2 2

compilation warnings
by Vieri 05 Oct '12

05 Oct '12

Hi, On a Gentoo Linux system I'm getting the following compile-time warnings (kamailio-3.3.1 ebuild): * python_msgobj.c:529:5: warning: dereferencing type-punned pointer will break strict-aliasing rules * ld_con.c:241:4: warning: implicit declaration of function 'ldap_simple_bind_s' * perl.c:340:2: warning: null argument where non-null required (argument 2) * perl.c:340:2: warning: null argument where non-null required (argument 1) Hopefully I can just ignore these messages. Thanks, Vieri

3 3

kamailio 3.3 sipcapture and Homer Capture Agent 4
by Juha Heinanen 05 Oct '12

05 Oct '12

is Homer Capture Agent 4 hep protocol incompatible with Kamalio 3.3 sipcapture module? in captagent.xml i set core_hep.conf version to 2, but looks like kamailio is not able to parse the messages: Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:145]: ERROR:parse_first_line: method not followed by SP Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:242]: ERROR:parse_first_line: bad message Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [parser/msg_parser.c:705]: ERROR: parse_msg: message=<#002#020#002#006#023ĳ��bg#012�bg#012#002�nP#037#035#010> Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [receive.c:146]: ERROR: receive_msg: parse_msg failed in kamailio config i have modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) -- juha

2 2

Does Kamailio support ECDHE ciphers for TLS?
by AJ 05 Oct '12

05 Oct '12

Hi, I have Kamailio 3.3.1 working fine with TLS 1.0, running on CentOS 6.3 However, now I want to use some ECDHE ciphersuites, for instance: ECDHE-ECDSA-AES128-SHA I have built an appropriate OpenSSL (1.0.0j) in CentOS to support this, and have generated the appropriate certs: openssl ecparam -out ca-privkey.pem -name secp256r1 -genkey -outform pem openssl req -x509 -new -key ca-privkey.pem -out ca_cert.pem -outform PEM -days 3650 openssl ecparam -out privkey.pem -name secp256r1 -genkey -outform pem openssl req -new -nodes -key privkey.pem -outform pem -out cert_req.pem openssl ca -keyfile ca-privkey.pem -cert ca_cert.pem -in cert_req.pem -out cert.pem However, when I try to connect from my phone I am always getting: Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: tls [tls_server.c:1190]: TLS accept:error:1408A0C1:SSL routines:SSL3_GET_CLIENT_HELLO:no shared cipher Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: <core> [tcp_read.c:1127]: ERROR: tcp_read_req: error reading If I use the OpenSSL s_server test-tool, I can connect from my phone just fine with these certs: openssl s_server -accept 8888 -cert cert.pem -key key.pem -pass stdin -CAfile calist.pem -cipher ECDHE-ECDSA-AES128-SHA Does Kamailio support ECDHE ciphersuites? If so, what am I doing wrong? Thanks! AJ

3 2

Kamailio question
by Jostein Aardal 05 Oct '12

05 Oct '12

Hi! I am a student at Oslo and Akershus University college of applied sciences. I'm working on a project where I am supposed to setup a kamailio server. Up until now I have been able to install kamailio and add a user. When I use a sip-client I am able to register and authenticate this user. And I am able to make calls between different users, as long as they both are registered on the same server. But when I try to make a call between users that are registered on different servers/proxies I fail to establish the call. When I look at the sip-messages in wireshark, I see that there is a message "407 proxy authentication required". I thought that enabling "multidomain" in kamailio.cfg would fix this problem, but it doesn't. What I have written in the kamailio.cfg-file is: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_ALIASDB #!define WITH_DEBUG yours Jostein Aardal Oslo and Akershus University College of Applied Sciences

3 2

hello i want to active international call
by mark shaw 05 Oct '12

05 Oct '12

hello i want to active international call in my account for important . radioyork2 thedept

2 1

Client info
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, I am trying to use Kamailio 3.3 configured as a presence server and rls server with integrated XCAP. I am trying to test the presence and rls functionality with clients such as Jitsi (doesn't seem to support rls and doesn't let me specify presence rules per contact) and Boghe (doesn't seem to be behaving as expected!). Anyone recommend other clients that work for the functionality I described above. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio LCR - From Uri
by Fatima Chahrour~Vanrise Support 04 Oct '12

04 Oct '12

Dears, Am able to route call using LCR module from anywhere. I need to use certain option in the lcr tables in order to specify the ip of which calls is generated. I suppose that the From_uri column in LCR module meets my requirement, yet am not able to make it work and confused after reading several docs. My question is in what form should the from_uri column be? Is the ip enough? Are the Load params used below missing? modparam("lcr", "db_url","mysql://openser:openserrw@localhost/openser") modparam("lcr","lcr_gw_table","lcr_gw") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr","lcr_id_avp","$avp(s:lcr_id_avp)") What should I change in the routing script to consider the from_uri? if (!load_gws("1")) { sl_send_reply("503", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} Thanks in advance, F Chahrour

3 3

sipcapture unsupported family
by Juha Heinanen 04 Oct '12

04 Oct '12

after adding a few module params modparam("sipcapture", "db_url", "MYSQL_SIP_PROXY_USAGE_URL") modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "capture_node", "EXTERNAL_HOSTNAME") i restarted my sip proxy and started to get these kind of errors to syslog: Oct 4 18:09:02 siika /usr/sbin/sip-proxy[22914]: ERROR: sipcapture [sipcapture.c:733]: ERROR: sipcapture:hep_msg_received: unsupported family [71] why they appear and how to turn them off? -- juha

3 4

Presence with RLS and integrated XCAP on Kamailio 3.3
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, Does anyone have presence working on Kamailio 3.3 with integrated XCAP. In theory my setup should work. I have two clients: Client A: Registers with the presence server Subscribe 1 - event = reg Subscribe 2 - event = presence.winfo Subscribe 3 - event = presence Publishes its status I see entries in the foll databases for A: Active Watchers DB: presence, presence.winfo, and reg event entries Watchers DB: No entries Presentity: presence, reg, and presence???? entries PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 3 entries - rls services, resource list, and pres rules Client B: Registers with the presence server Subscribe 1 - event = presence.winfo Subscribe 2 - event = presence Subscribe 3 - event = message-summary Publishes its status I see entries in the foll databases for B: Active Watchers DB: message-summary, presence.winfo Watchers DB: 1 entry - with B as the presentity and A as the watcher Presentity: reg, presence PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 2 entries resource-lists, pres rules My first question is - Is this all correct or is something going missing? Now the real problem: When Client A adds Client B to it's contact list, on Client A I see the following message: Client B Package (presence), Pending (Subscribe). This information from what I gather is coming back from the periodic notifies for the server because of the presence.winfo event. However, the notify going out to Client B does not have any indication that Client A is a watcher and needs to be authorized. This should be the case per RFC 3857. Excerpts from notifies to client A and client B: Client A NOTIFY sip:8475551001@10.51.2.181:58150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bKdcef.e5df11e.0 To: sip:8475551001@10.50.251.12;tag=1988526 From: sip:8475551001@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-c52c CSeq: 3 NOTIFY Call-ID: 19607474-addd-0ffb-a9da-031235052e9d Content-Length: 347 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3180 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="partial"> <watcher-list resource="sip:8475551001@10.50.251.12" package="presence"> <watcher id="9e65d804dbc4fc269c5f1d2f6eae0650(a)0.0.0.0" event="subscribe" status="pending">sip:8475551002@10.50.251.12</watcher> </watcher-list> </watcherinfo> Client B: NOTIFY sip:8475551002@10.50.0.113:5060;transport=udp;registering_acc=10_50_251_12 SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bK7f71.6fca9e81.0 To: sip:8475551002@10.50.251.12;tag=a30db42f From: sip:8475551002@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6692 CSeq: 3 NOTIFY Call-ID: f71666432f236f38c7c4dc467b4483e0(a)0.0.0.0 Content-Length: 195 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3600 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="full"> <watcher-list resource="sip:8475551002@10.50.251.12" package="presence"/> </watcherinfo> Also the frequency of notifies to client B is a lot less for event: presence.winfo. Any help is greatly appreciated since I've been trying to get presence to work for a few weeks now and keep running into new problems. Thanks, Sangeeta -- Sangeeta Shah

1 0

sip_capture schema?
by Juha Heinanen 04 Oct '12

04 Oct '12

i couldn't find in the source directory database schema for sip_capture table. why it is not included? also, looks like sipcapture module README is missing many of module's params. is this module still supported or has it become obsolete? -- juha

3 9

Kamailio - can't write any logs to file
by Nord7 04 Oct '12

04 Oct '12

Hello everyone. I have installed: CentOS6, Kamailio 3.3, rsyslogd Part of my Kamailio config: debug=9 log_stderror=yes memdbg=9 memlog=9 log_facility=LOG_LOCAL0 loaded modules: debugger.so modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_facility", "LOG_LOCAL0") modparam("debugger", "log_prefix", "from-debugger-with-love:") xlog.so modparam("xlog", "buf_size", 8192) modparam("xlog", "log_facility", "LOG_LOCAL0") rsyslog.conf default config with two added rows: # Save log messages with LOG_LOCAL0 in /var/log/kamailio local0.* /var/log/kamailio rsyslogd and kamailio restarted. When i run command "kamailio -ddddddd" i see debug all my xlog("L_INFO","blabla\n") or xlog("LOG_LOCAL0","L_INFO","blabla\n" ); messages. When i run "kamctl start" i don't have any logs from kamailio in my system. I'm stuck, any suggestions, will be helpful -- Best regards, Kirill Ushakov

2 2

Re: [SR-Users] Error when client tries to put XCAP resource list
by Daniel-Constantin Mierla 03 Oct '12

03 Oct '12

Hello, do not reply privately to discussions on the mailing list. It is important that the thread has a conclusion for the people searching in archive for the same topic. Cheers, Daniel On 10/3/12 10:13 PM, Sangeeta Shah wrote: > Thanks for the response. I was able to figure it out and now it works. > > Appreciate your response. > > On Wed, Oct 3, 2012 at 3:02 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> the http uri is available in available $hu. You can print it with xlog and >> be sure you give it as parameter to xcap server function. >> >> A basic example in the readme shows it: >> * >> http://kamailio.org/docs/modules/stable/modules_k/xcap_server.html#id2532050 >> >> Cheers, >> Daniel >> >> >> >> On 10/3/12 8:34 PM, Sangeeta Shah wrote: >>> Hello All, >>> I am using Kamailio 3.3 with Presence/RLS and integrated XCAP. When >>> my client tries to "PUT" the resource list for the user, the server is >>> returning a 500 Server Error response. Attached is the TCP trace for >>> the put and the corresponding response from the server. I also see the >>> following error in the server logs: >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> <script>: ===== xhttp: put received sip:8475551001@10.50.251.12 >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> xcap_server [xcap_misc.c:109]: invalid http uri - shorter than >>> xcap-root >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> xcap_server [xcap_server.c:560]: cannot parse xcap uri [0] >>> >>> Maybe it's a client/server configuration issue. What is the expected >>> http uri. How do I print out the received HTTP URI And the XCAP root. >>> >>> Any debugging help is appreciated. >>> >>> TCP trace: >>> >>> PUT /xcap-root/resource-lists/users/8475551001/index HTTP/1.1 >>> >>> Host: 10.50.251.12:5060 >>> >>> Content-Length: 2037 >>> >>> Connection: Keep-Alive >>> >>> User-Agent: XDM-client/OMA1.1 >>> >>> X-3GPP-Intended-Identity: 8475551001 >>> >>> Content-Type: application/resource-lists+xml >>> >>> >>> >>> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> >>> >>> <list name="rcs"> >>> >>> <display-name>All Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <entry uri="8475551001" /> >>> >>> </list> >>> >>> <list name="rcs_blockedcontacts"> >>> >>> <display-name>Blocked Contacts</display-name> >>> >>> </list> >>> >>> <list name="rcs_revokedcontacts"> >>> >>> <display-name>Revoked Contacts</display-name> >>> >>> </list> >>> >>> <list name="oma_allcontacts"> >>> >>> <display-name>OMA All Contacts</display-name> >>> >>> </list> >>> >>> <list name="oma_blockedcontacts"> >>> >>> <display-name>OMA Blocked Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_buddylist"> >>> >>> <display-name>OMA BuddyList</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_grantedcontacts"> >>> >>> <display-name>OMA Granted Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_pocbuddylist"> >>> >>> <display-name>OMA POC BuddyList</display-name> >>> >>> </list> >>> >>> <list name="doubango"> >>> >>> <display-name>My Default Contacts</display-name> >>> >>> </list> >>> >>> </resource-lists> >>> >>> HTTP/1.1 500 Server error >>> >>> Via: SIP/2.0/TCP 10.51.2.181:49889 >>> >>> Server: kamailio (3.3.1 (x86_64/linux)) >>> >>> Content-Length: 0 >>> >>> Thanks, >>> Sangeeta >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >> http://asipto.com/u/katu >> > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Error when client tries to put XCAP resource list
by Sangeeta Shah 03 Oct '12

03 Oct '12

Hello All, I am using Kamailio 3.3 with Presence/RLS and integrated XCAP. When my client tries to "PUT" the resource list for the user, the server is returning a 500 Server Error response. Attached is the TCP trace for the put and the corresponding response from the server. I also see the following error in the server logs: Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: <script>: ===== xhttp: put received sip:8475551001@10.50.251.12 Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: xcap_server [xcap_misc.c:109]: invalid http uri - shorter than xcap-root Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: xcap_server [xcap_server.c:560]: cannot parse xcap uri [0] Maybe it's a client/server configuration issue. What is the expected http uri. How do I print out the received HTTP URI And the XCAP root. Any debugging help is appreciated. TCP trace: PUT /xcap-root/resource-lists/users/8475551001/index HTTP/1.1 Host: 10.50.251.12:5060 Content-Length: 2037 Connection: Keep-Alive User-Agent: XDM-client/OMA1.1 X-3GPP-Intended-Identity: 8475551001 Content-Type: application/resource-lists+xml <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="rcs"> <display-name>All Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <entry uri="8475551001" /> </list> <list name="rcs_blockedcontacts"> <display-name>Blocked Contacts</display-name> </list> <list name="rcs_revokedcontacts"> <display-name>Revoked Contacts</display-name> </list> <list name="oma_allcontacts"> <display-name>OMA All Contacts</display-name> </list> <list name="oma_blockedcontacts"> <display-name>OMA Blocked Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_buddylist"> <display-name>OMA BuddyList</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_grantedcontacts"> <display-name>OMA Granted Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_pocbuddylist"> <display-name>OMA POC BuddyList</display-name> </list> <list name="doubango"> <display-name>My Default Contacts</display-name> </list> </resource-lists> HTTP/1.1 500 Server error Via: SIP/2.0/TCP 10.51.2.181:49889 Server: kamailio (3.3.1 (x86_64/linux)) Content-Length: 0 Thanks, Sangeeta -- Sangeeta Shah

2 1

Re: [SR-Users] Handling Tel URIs
by Kamal Palei 03 Oct '12

03 Oct '12

Hi Daniel It worked fine. I added the below code. if(is_method("PUBLISH")) { loose_route(); t_relay(); exit; } Thanks Kamal On Wed, Oct 3, 2012 at 5:58 PM, Kamal Palei <palei.kamal(a)gmail.com> wrote: > Thanks Daniel , let me try this out. > > Regards > Kamal > > On Wed, Oct 3, 2012 at 5:33 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> do you have a Route header also for kamailio proxy? If yes, try to do >> loose_route() and then t_relay(). >> >> If not, get the address from Route in $du and relay: >> $du = $(hdr(Route){nameaddr.uri}); >> remove_hf("Route"); >> t_relay(); >> exit; >> >> Cheers, >> Daniel >> >> >> On 10/3/12 1:46 PM, Kamal Palei wrote: >> >> Hi Daniel >> Thanks for your response. >> >> I have a message as below. >> >> PUBLISH tel:+915346586495469 <+915346586495469> SIP/2.0 >> Via: >> To: >> From: >> Route: <sip:main_proxy_ip> >> .. >> .. >> This message comes from phone to Kamailio proxy. >> I just need Kamailio proxy just forward this message to main proxy >> without modifying tel uri present in request uri. >> >> Can this be acheived by tel2sip() api. >> >> Thanks >> Kamal >> >> >> On Wed, Oct 3, 2012 at 4:37 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 10/3/12 12:36 PM, Kamal Palei wrote: >>> >>> Changing the subject line >>> >>> On Wed, Oct 3, 2012 at 3:54 PM, Kamal Palei <palei.kamal(a)gmail.com>wrote: >>> >>>> Hi All >>>> I am using Kamailio - 3.1.5 version. >>>> >>>> When a message of tel URI comes to Kamailio, Kamailio is not forwarding >>>> the message to next proxy. >>>> If is returning "478 Unresolved response" . >>>> >>>> Can somebody give me some pointer how do I handle this situation? >>>> >>> kamailio cannot forward when there is tel uri -- it requires a domain >>> part (hostname/ip) to know where to send the requests. >>> >>> See this function that should help you convert the tel uri to sip uri: >>> http://kamailio.org/docs/modules/stable/modules_k/siputils.html#id2494464 >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >>> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >> >> >

1 0

Re: [SR-Users] Handling Tel URIs
by Kamal Palei 03 Oct '12

03 Oct '12

Changing the subject line On Wed, Oct 3, 2012 at 3:54 PM, Kamal Palei <palei.kamal(a)gmail.com> wrote: > Hi All > I am using Kamailio - 3.1.5 version. > > When a message of tel URI comes to Kamailio, Kamailio is not forwarding > the message to next proxy. > If is returning "478 Unresolved response" . > > Can somebody give me some pointer how do I handle this situation? > > Thanks > Kamal > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

2 3

Kamailio CDR logging to DB
by Tayeb Meftah 03 Oct '12

03 Oct '12

hi everyone, can someone show me how to save cdr's into db ? i enabled acc in the default config but still no cdr in the acc table thank you -- Tayeb Meftah General director Voice of the blind ™ broadcast freedom http://www.vobradio.org/ Phones: Algeria: +213-660-347746 UK, London: +442032983627 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

2 1

Locate UA in usrloc table
by Vitaliy Aleksandrov 03 Oct '12

03 Oct '12

Hi all. Is there any easy method to find if somebody with $si:$sp is registered (usrloc->received.ip == $si, usrloc->received.port == $sp) ? reg_fetch_contacts uses uri as search key, but in my case I don't have it. The only way I know is to write a module that will use "get_all_ucontacts" function exported by usrloc module in its API.

2 1

FreeSWITCH | Kamailio MWI forwarding
by Tayeb Meftah 03 Oct '12

03 Oct '12

hello we're using Kamailio / freeswitch freeswitch is the Voicemail server now. please how to forward MWI bethwan kamailio and Freeswitch? if user 1000 have a vm i want kamailio to fetch it from FS and advertise it to the subscriber thank you -- Tayeb Meftah General director Voice of the blind ™ broadcast freedom http://www.vobradio.org/ Phones: Algeria: +213-660-347746 UK, London: +442032983627 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

2 1

WARNING:fm_free at stop
by Juha Heinanen 03 Oct '12

03 Oct '12

at some point i have started to get these kind of warning when i stop my sip proxy: --text follows this line-- Oct 1 10:23:01 siika /usr/sbin/sip-proxy[16970]: WARNING: <core> [mem/f_malloc.c:512]: WARNING:fm_free: free(0) called any idea where they come from or how i could found out where they come from? -- juha

2 20

(no subject)
by Kamal Palei 03 Oct '12

03 Oct '12

Hi All I am using Kamailio - 3.1.5 version. When a message of tel URI comes to Kamailio, Kamailio is not forwarding the message to next proxy. If is returning "478 Unresolved response" . Can somebody give me some pointer how do I handle this situation? Thanks Kamal

1 0

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

I will check. The problem is that this installation is exotic and deviant from my usual order of operations, which creates further confusion. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: On 10/2/12 9:45 PM, Alex Balashov wrote: Perhaps the source of my misunderstanding is something to do with branching. The actual case I am troubleshooting is a 503 generated via t_reply out of a failure route. The complaint is that this 503 contains a modified From header, exposed to the originator of the call. If you created the transaction after calling msg_apply_changes(), then I am quite sure it is how it worked all the time. Try with an older version and see. Cheers, Daniel -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Those originated locally were never using the original value if you called msg apply changes -- you can try with an older version, if it does that, then it is a bug somewhere :-). Btw, if you called msg apply changes after creating the transaction, then SL is out of scope and tm local reply is using the initial values, but that means the changes are only for that branch, a failure route will get the initial request content. Cheers, Daniel On 10/2/12 9:30 PM, Alex Balashov wrote: The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

Perhaps the source of my misunderstanding is something to do with branching. The actual case I am troubleshooting is a 503 generated via t_reply out of a failure route. The complaint is that this 503 contains a modified From header, exposed to the originator of the call. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:Those originated locally were never using the original value if you called msg apply changes -- you can try with an older version, if it does that, then it is a bug somewhere :-). Btw, if you called msg apply changes after creating the transaction, then SL is out of scope and tm local reply is using the initial values, but that means the changes are only for that branch, a failure route will get the initial request content. Cheers, Daniel On 10/2/12 9:30 PM, Alex Balashov wrote: The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Alex Balashov <abalashov(a)evaristesys.com> wrote:Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

Kamailio project booth at Astricon 2012
by Daniel-Constantin Mierla 02 Oct '12

02 Oct '12

Hello, Kamailio project will participate at Astricon 2012 with its own booth in the open source area, courtesy of our Asterisk friends organizing the show. The event takes place in Atlanta, GA, USA, between Oct 23-25, 2012: * http://www.astricon.net/2012/default.aspx Special thanks to community members JR Richardson, Nir Simionovich and Eric Klein, which coordinate the activity at the booth. I made a news about it at: * http://www.kamailio.org/w/2012/10/kamailio-booth-at-astricon12/ If you are interested in joining the team at the booth and present how you use Kamailio, reply to me in order discuss further. If you participate at Astricon, stop by booth #20 to chat about use cases of Kamailio and what is new around the project. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Kamailio - VoIP dinner in Berlin, next Monday, Oct 8, 2012
by Daniel-Constantin Mierla 02 Oct '12

02 Oct '12

Hello, we're planning a dinner in Berlin, next Monday, Oct 8, 2012 - exact time and place to be decided soon. Besides me, among people interested to participate are Alex Balashov, Andrei Pelinescu-Onciul, Jiri Kuthan and Raphael Coeffic. If anyone here wants to participate, reply to count the heads for reservation. It's the usual german rule, everyone pays for him/herself. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

XCAP configuration with Presence Server and Resource lists not working on Kamailio 3.3.1
by Sangeeta Shah 01 Oct '12

01 Oct '12

Hello All, I am trying to get the presence module on Kamailio 3.3.1 working with integrated XCAP and resource lists. The clients I am using to test are: Jitsi and Boghe. I am following the configuration as specified by the respective module documentation: http://kamailio.org/docs/modules/devel/modules_k/xcap_server.html http://kb.asipto.com/kamailio:presence:k31-made-simple http://www.kamailio.org/dokuwiki/doku.php/presence:presence-module and http://kamailio.org/docs/modules/devel/modules_k/rls.html The server is up and running but it doesn't seem like it's working for either one of the clients. I checked the archives and someone had a similar issue in Aug and switched back to releases 3.2/3.1. Has anything been resolved? Are there any changes that need to be made to the documentation for 3.3 to support this functionality. Any help is appreciated. If tcpdumps for the respective clients are desired I can provide them. Thanks, Sangeeta -- Sangeeta Shah

2 2

Error starting kamailio with dialog module usage
by Aft nix 01 Oct '12

01 Oct '12

Hi, After implementing a "Call limit" application with the following config: #!ifdef WITH_CALL_LIMIT modparam("dialog","enable_stats",1) modparam("dialog","dlg_flag", 4) modparam("dialog","hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) #!endif request_route { ................ #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$rU", "var(100)"); if ($var(100) >= 5 ) { xlog("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$ru"); if (get_profile_size("callquota", "$var(100)")) { xlog("DEBUG: there are $var(100) total calls for $rU\n"); } } #!endif ............... } Now it passes cleanly when i issue : $kamailio -c but it fails to start with following info: Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:507 Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:523 line 507 : get_profile_size("callquota", "$rU", "var(100)"); Thanks in advance -- -aft

2 2

kamailio core at qm_status
by Jijo 01 Oct '12

01 Oct '12

Hi All, I'm observing a core intermittently at "qm_status (qm=0x786cd000) at mem/q_malloc.c:763" for kamailio version 3.1.0 looking at the backtrace this is occurring while doing a tcp buffer overrun. Please have a look and let me know if anyone observed this issue or how can we debug it. Please find the backtrace and kamailio version osbprod-V2R0:~ # /usr/sbin/kamailio -V version: kamailio 3.1.0 (i386/linux) 21a375 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 15MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 21a375 compiled on 12:38:44 Apr 26 2012 with gcc 4.5.0 #0 qm_status (qm=0x786cd000) at mem/q_malloc.c:763 #1 0x0819c0f3 in qm_debug_frag (qm=0x786cd000, f=<value optimized out>) at mem/q_malloc.c:167 #2 0x0819c352 in qm_free (qm=0x786cd000, p=0x7a6cb6ac, file=0x821cce2 "<core>: tcp_main.c", func=0x822285f "wbufq_run", line=880) at mem/q_malloc.c:457 #3 0x0815f686 in wbufq_run (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:880 #4 handle_tcpconn_ev (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:4141 #5 0x08169bf2 in io_wait_loop_epoll () at io_wait.h:1092 #6 tcp_main_loop () at tcp_main.c:4606 #7 0x080b0404 in main_loop () at main.c:1655 #8 0x080b1f84 in main (argc=9, argv=0xbfb4c9f4) at main.c:2446 (gdb) l 758 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ", 759 "dumping free list stats :\n"); 760 for(h=0,i=0;h<QM_HASH_SIZE;h++){ 761 unused=0; 762 for (f=qm->free_hash[h].head.u.nxt_free,j=0; 763 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++){ 764 if (!FRAG_WAS_USED(f)){ 765 unused++; 766 #ifdef DBG_QM_MALLOC 767 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ",

4 7

How to force tcp connection reuse for NOTIFY
by Owen Lynch 01 Oct '12

01 Oct '12

Hi, I want to configure Kamailio so that it sends a NOTIFY using the tcp connection already established by the client SUBSCRIBE request. The aim is for clients to establish secure TLS connections to Kamailio when they first register or subscribe, then for kamailio to reuse these connections for all further NOTIFY/INVITE requests. I thought that adding force_tcp_alias() in the route script would accomplish this, but it doesn't. The NOTIFY always uses the ip:port tuple specified in the SUBSCRIBE Contact header. Can anyone shed any light on this? Thanks, Owen Lynch

2 2

Registration Authentication Error
by Nathaniel L Keeling III 30 Sep '12

30 Sep '12

Hello, we are testing the new version of Kamailio, 3.3 and are getting authentication error when trying to register a sip account. We have a db backend in which we used kamctl to add the user to the db. We are using the kamailio.cfg script created during the install with changes for our environment. Is there some other changes required besides the normal. I can see in the logs where everything looks ok except for when it compares the calculated values. I have doubled check the subscriber table and password but to no avail. 11(28548) DEBUG: db_postgres [km_dbase.c:393]: PQclear(1008b8560) result set 11(28548) DEBUG: auth_db [authorize.c:124]: HA1 string calculated: e13d569284e76a33ca63e4d6203f844a 11(28548) DEBUG: auth [api.c:211]: check_response: Our result = '18210b5da2b3d27e6ba50977072599ea' 11(28548) DEBUG: auth [api.c:221]: check_response: Authorization failed Thanks Nathaniel

5 18

Kamailio 3.3 presence with resource lists and integrated XCAP
by Sangeeta Shah 28 Sep '12

28 Sep '12

Hello All, I am trying to configure Kamailio v 3.3 to serve as a presence server with resource lists and integrated XCAP. I followed the respective module documentation and am trying to test with the Boghe IMS client. I just can't get it to work! I think I still don't have everything I need in the config file to handle the xcap documents. Attached is the debug log from the server. If someone has had luck setting up the presence server with integrated xcap and resource lists please take a few minutes to take a look at the log file and see if you can help me out. I am also posting the xcap portion of the config file: #!ifdef WITH_XCAPSRV ##!define WITH_XHTTPAUTH event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); #!ifdef WITH_XHTTPAUTH if (!www_authorize("10.50.251.12", "subscriber")) { www_challenge("10.50.251.12", "0"); exit; } #!endif if($hu=~"^/xcap-root/") { set_reply_close(); set_reply_no_connect(); # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if($xcapuri(u=>auid)=="xcap-caps") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>pres-rules</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "ok", "application/xcap-caps+xml", "$var(xbody)"); exit; } #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents if ($au!=$(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } #!endif switch($rm) { case "PUT": #xcaps_put("$var(uri)", "$hu", "$rb"); #if($xcapuri(u=>auid)=~"pres-rules") #{ # xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); # pres_update_watchers("$var(uri)", "presence"); # pres_refresh_watchers("$var(uri)", "presence", 1); #} xcaps_put("$var(uri)", "$var(doc_uri)", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid)=~"resource-lists" || $xcapuri(u=>auid)=~"rls-services") { rls_update_subs("$var(uri)", "presence"); } exit; break; case "GET": xlog("===== xhttp: get $var(uri) => $hu\n"); xcaps_get("$var(uri)", "$hu"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$hu"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; } } # http ops xhttp_reply("200", "ok", "text/html", "<html><body>OK: $si:$sp</body></html>"); exit; } -- Sangeeta Shah

1 0

DB accounting missing if t_newtran() is called explicitly
by Andrew Pogrebennyk 28 Sep '12

28 Sep '12

Hi, I have found recently that in order to detect retransmits I have to create a transaction explicitly when the request comes in: force_rport(); if(!t_check_trans()) t_newtran(); sl_send_reply("100", "Trying"); xlog("L_INFO", "New request - $ci\n"); it appears like there are carriers or UAs that do not honor the T1 retransmission interval retransmit the INVITE sooner than proxy creates a transaction in t_relay(). And since we are counting concurrent calls, we count the same call multiple times, which is not good. But with this patch we've faced another sporadic problem - if the transaction is created beforehand the accounting record is lost.. we use acc_db mode and set flag to account the transaction. And there are no errors in kamailio log but no insert into acc in mysql binlog either. I wasn't successful reproducing it in the lab systems with identical setup. Is anybody here perhaps aware of some limitation in acc module or callbacks which makes a transaction created beforehand not accountable? On a related note, it could make sense to create a transaction implicitly if dlg_manage() is called to avoid counting same call many times, I just don't know yet how common this issue is in real life.

2 4

SRV record in via header not resolving.
by Jijo 28 Sep '12

28 Sep '12

Hello, I'm observing that Via Header domain name of type SRV is resolving as DNS A. I looked at the code and it does support only A. It is basically setting to default SIP port check_via_address which is causing DNS A query. Is there any specific reason to do so? I changed the code to support SRV as below and working fine. static int check_via_address(struct ip_addr* ip, str *name, unsigned short port, *short proto*, int resolver) { : : //if (port==0) port=SIP_PORT; if (resolver&DO_DNS){ DBG("check_via_address: doing dns lookup\n"); /* try all names ips */ he=sip_resolvehost(name, &port, *(char *) &proto*); /* don't use naptr */ if (he && ip->af==he->h_addrtype){ for(i=0;he && he->h_addr_list[i];i++){ if ( memcmp(&he->h_addr_list[i], ip->u.addr, ip->len)==0) return 0; } } } : : : } /* check if IP address in Via != source IP address of signaling */ int received_test( struct sip_msg *msg ) { int rcvd; rcvd=msg->via1->received || check_via_address(&msg->rcv.src_ip, &msg->via1->host, msg->via1->port,* msg->via1->proto*, received_dns); return rcvd; } Thanks Jijo

2 2

Kamailio sip domain
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

Hi, What is the sip domain in kamctlrc file? Can I change it whatever I want? Is it must be a valid sip domain? Can I give a name like "mysipdomain.com", but there is no server with this name?

2 1

invalid version 6 for table starting from git rev 78dae896127ce6762e3fa7c2541e1b5f9b8a9023
by siklub 28 Sep '12

28 Sep '12

Hi All, This is what i have done. * get kamailio from git git clone git://git.sip-router.org/sip-router * Installing kamailio from working rev A * destroy the database * update kamailio code from git * installing kamailio from not working rev B * create the database again (kamdbctl create) * starting kamailio This is the error that i got. ep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: <core> [db.c:440]: invalid version 6 for table location found, expected 5 (check table structure and table "version") Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: usrloc [dlist.c:640]: error during table version check. Sep 28 13:52:04 horse /usr/local/sbin/kamailio[28329]: ERROR: registrar [reg_mod.c:484]: failed to register domain I try to find last working rev Last working rev is (rev A) : ca551f7cb3770a08832758e543587415b3c6d80d (2012-09-11) First not working rev is (rev B) : 78dae896127ce6762e3fa7c2541e1b5f9b8a9023 There is no special reason for destroying the database, i'm just playing with kamailio from https://www.kamailio.org/wiki/install/devel/git, it seems that i have done right procedure. Am i missed something? Thanks

2 2

dispatching xcap requests
by Juha Heinanen 28 Sep '12

28 Sep '12

is it possible to use kamailio based dispatcher to dispatch also xcap requests to one or more kamailio instances that serve them, i.e., is it possible to call t_relay in event_route [xhttp:request]? -- juha

3 2

Kamailio sip user authentication
by Burak ÖZTÜRK 28 Sep '12

28 Sep '12

I installed kamailio from this link: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git And I used Linphone and/or CSipSimple android apps as sip clients. And if I give the correct sip server name (usually server ip address), kamailio does not check the sip username and password and sip clients can register. In short, kamailio does not have authentication machanism for sip clients. Is it true or I'm doing something wrong? I searched the web and saw an authentication module called auth_identity. Is it a module to make authentication and if yes, how to activate it? Thanks Regards

2 1

non-local socket info message
by Juha Heinanen 27 Sep '12

27 Sep '12

when i do registrar save in multiple registrar setup, i get this kind of info message to syslog when socket field value in location table does not match local socket of the registrar: INFO: usrloc [udomain.c:321]: non-local socket <tcp:192.98.103.2:5070>...ignoring looks like location table entries are updated no matter of the message. what does "ignoring" mean? in my case i don't use socket field value for anything, because requests to contacts are always routed to dispatcher. would it make sense to have a module param that tells usrloc to skip handling of socket info? -- juha

1 0

help with kamailio crash
by Vicente Hernando 26 Sep '12

26 Sep '12

Hello, we are using a kamailio 3.3.1 version here and suddenly it crashed with a core. Any point about how to debug this will be appreciated! # /usr/local/kamailio/sbin/kamailio -V version: kamailio 3.3.1 (x86_64/linux) 91e8cb flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 91e8cb compiled on 04:59:57 Sep 13 2012 with gcc 4.6.3 (gdb) bt full #0 0x00007f242514e445 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f2425151bab in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x00000000004740b2 in sig_alarm_abort (signo=<optimized out>) at main.c:695 No locals. #3 <signal handler called> No symbol table info available. #4 0x00007f24252066d7 in syscall () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000542c58 in futex_get (lock=0x7f23e1ceb7a8) at mem/../futexlock.h:110 i = <optimized out> #6 timer_del_safe (tl=0x7f23e1d378e0) at timer.c:729 ret = -1 #7 0x0000000000447ab6 in destroy_dns_cache () at dns_cache.c:245 No locals. #8 0x0000000000474f95 in cleanup (show_status=1) at main.c:572 memlog = <optimized out> #9 0x0000000000475969 in shutdown_children (show_status=1, sig=15) at main.c:712 No locals. #10 0x00000000004767ac in handle_sigs () at main.c:803 chld = <optimized out> chld_status = 139 memlog = <optimized out> #11 0x0000000000478ed5 in main_loop () at main.c:1762 i = <optimized out> pid = <optimized out> si = <optimized out> si_desc = "udp receiver child=5 sock=79.170.68.187:5060\000\177", '\000' <repeats 18 times>, "\001\000\000\000\000\000\000\000\000\000\300>\000\000\000\000\001\000\000\000#\177\000\000Q\251U\000\000\000\000\000\020p\321$$\177\000\000\b\000\000\000\000\000\000\000x\264\316\341#\177\000\000\000gE\345\322\024P\027" nrprocs = <optimized out> #12 0x000000000041add2 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2546 cfg_stream = <optimized out> c = <optimized out> r = <optimized out> tmp = 0x7fff551d8f24 "" ---Type <return> to continue, or q <return> to quit--- tmp_len = 1427999204 options = 0x5df520 ":f:cm:M:dVIhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 603487729 rfd = 0 debug_save = <optimized out> debug_flag = <optimized out> dont_fork_cnt = <optimized out> n_lst = 0x0 p = <optimized out> Thanks in advance, Vicente.

2 1

Question about CPU
by Ricardo Martinez 26 Sep '12

26 Sep '12

Hello. Is normal this output from the “top” command?! I have a kamailio with 16 children receiving calls at rate of 30cps and using the dialog module to timeout the calls : This is the top command : Tasks: 168 total, 2 running, 166 sleeping, 0 stopped, 0 zombie Cpu0 : 56.0%us, 0.7%sy, 0.0%ni, 43.4%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu1 : 6.3%us, 3.0%sy, 0.0%ni, 87.8%id, 0.0%wa, 0.7%hi, 2.3%si, 0.0%st Cpu2 : 0.3%us, 0.3%sy, 0.0%ni, 99.3%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Cpu3 : 0.7%us, 0.3%sy, 0.0%ni, 99.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 3924912k total, 535700k used, 3389212k free, 20236k buffers Swap: 6291448k total, 0k used, 6291448k free, 226120k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND *2566 root 20 0 1163m 27m 23m R 56.2 0.7 2:08.32 kamailio * 1103 root 20 0 249m 1656 924 S 2.7 0.0 1:32.89 rsyslogd 2525 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.93 kamailio 2527 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.11 kamailio 2528 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.89 kamailio 2529 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.13 kamailio 2535 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.86 kamailio 2537 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.85 kamailio 2538 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.17 kamailio 2541 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.91 kamailio 2542 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.07 kamailio 2543 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.12 kamailio 2544 root 20 0 1163m 27m 23m S 0.7 0.7 0:02.64 kamailio 2549 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.10 kamailio 2551 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.14 kamailio 2552 root 20 0 1163m 27m 23m S 0.7 0.7 0:03.00 kamailio 2495 root 20 0 15088 1240 900 R 0.3 0.0 0:04.10 top 2526 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.83 kamailio 2532 root 20 0 1163m 27m 23m S 0.3 0.7 0:02.87 kamailio 1 root 20 0 19396 1500 1192 S 0.0 0.0 0:02.86 init 2 root 20 0 0 0 0 S 0.0 0.0 0:00.01 kthreadd 3 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 4 root 20 0 0 0 0 S 0.0 0.0 0:00.00 ksoftirqd/0 5 root RT 0 0 0 0 S 0.0 0.0 0:00.00 migration/0 6 root RT 0 0 0 0 S 0.0 0.0 0:00.00 watchdog/0 The question is : I always see the process PID : 2566 using more CPU than the rest of the childrens. Is this normal? Regards, Ricardo Martinez-.

2 3

LDAP user authentication
by martian＠centrum.sk 26 Sep '12

26 Sep '12

Hi. I have been looking into LDAP way of authenticating users. Reading this guide http://www.kamailio.org/dokuwiki/doku.php/tutorials:kamailio31-auth-ldap and reading the AUTH and LDAP module documentations, it seems to me that currently you can bind to LDAP (using some service account for example) and perform the SEARCH operation for data only. Therefore you need to retrieve user login and password from the LDAP db and than authenticate the user in Kamailio. My question is: What is you can't simply retrieve the password from LDAP db?? Is it possible to do a BIND operation to LDAP, using login name and password provided by user in REGISTER message? (this means not using the ones specified in the external ldap config file). BIND operation kind of authenticates the user. So theoreticaly, if LDAP binding authentication succeeds, the user is trusted and can be replied with 200 OK. This in fact means: using bind operation instead of search operation when a REGISTER message (with Authorization header) arrives. Any opinions on corectness of this approach are welcome, along with clarifying the possibility to do this. Thanks in advance. Martin

1 0

Audiocide
by Cong Le Kha 26 Sep '12

26 Sep '12

Dear Kamailio I have got a problem as you with Audiocode SIP/ SECurity/ MGmt/ DebugRecording/ ControlProtocol/ CONFiguration/ IPNetworking/ TPApp/ BSP/ PING SHow />NOTIC:( sip_stack)(185 ) AcSIPParser: Problem in SIP Message Headers RECOV:( sip_stack)(186 ) !! [ERROR] AcSIPParser: Parse Error. No appropriate delimiter found in multipart body RECOV:( sip_stack)(187 ) !! [ERROR] Message type: INVITE RECOV:( sip_stack)(188 ) !! [ERROR] Source header: RECOV:( sip_stack)(189 ) !! [ERROR] Line: 41. Column: 1 Can you help me?

3 2

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

5 5

Kamailio and SylkServer
by Gary Shergill 25 Sep '12

25 Sep '12

Hi Kamailio Community, I was wondering how I would integrate Kamailio into SylkServer please? The SIP IM/presence part of Kamailio that is. When I was attempting to get the Presence and IM working I received the following reply; =================== > If you use sylkserver, then you can just forward the >INVITE to sylkserver, which has an MSRP relay integrated, =================== I'm not entirely sure what that means, but I found the following which I believe relates to it; http://sylkserver.ag-projects.com/projects/sylkserver/wiki/Applications I'm not entirely sure if; a) this is the right thing I'm looking at, b) where I would add these lines to Kamailio. Thanks for any help you can provide. Kind Regards, Gary Shergill

1 1

Asterisk and Dispatcher Module
by Ian French 25 Sep '12

25 Sep '12

Hi, I've been working my way throught this tutorial ( http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb) and all it working well thanks. In the other benifits section you mention that this configuration can be used with multiple instances of Asterisk. Could you elaborate a little on this?? Is this achieved using the Dispatcher module? I'm trying to get this working with two instances of Asterisk but I can figure out how to specifc multiple asterisk servers as oppossed to the single. asterisk.bindip = "192.168.178.23" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" Thanks in advance Ian

2 1

Re: [SR-Users] is kamilio support RFC5626 now ?
by Peter Dunkley 25 Sep '12

25 Sep '12

Hi, I have been working on adding full RFC5626 support to Kamailio in my spare time - but it is taking longer than I hoped because I haven't been able to spend as much time as I wanted working on it. There is a branch in git (pd/outbound) where I have put some of the module skeleton code, and I still hope to get this done and tested well before the end of the year (and certainly in time for Kamailo 3.4). Regards, Peter Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Starting kamailio. Error on topoh module
by Miguel Baptista 25 Sep '12

25 Sep '12

Hello, I am testing the "topology hiding" module (topoh) on kamailio-3.2.0 and every-time I start kamailio I get these error messages related with the topoh module: Sep 20 16:51:14 trd-sgw1 /usr/proxy/kamailio/sbin/kamailio[31768]: ERROR: topoh [th_mask.c:79]: +++ ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789.- Sep 20 16:51:14 trd-sgw1 /usr/proxy/kamailio/sbin/kamailio[31768]: ERROR: topoh [th_mask.c:80]: +++ y0GxFtSUVeTOMWzHPam.jKR93gCoNpkicrJBqIEhvYbnd175AfLl4u6wZs-X8Q2D I am only getting those errors when kamailio is starting. Kamailio runs fine and it hides my internal topology, on the SIP Headers, without any error message. I tried to add/modify/remove several topoh parameters but I am always getting those errors starting kamailio. Any idea why I getting it? How can I get rid of it? I don't feel comfortable to apply that configuration on my production servers knowing that kamailio outputs an error when starting. Thanks, - Miguel Baptista

2 3

Dialog Module : "inconsitent dlg timer data"
by Ricardo Martinez 25 Sep '12

25 Sep '12

Hello. Can someone explain to me why i’m getting this error : WARNING: dialog [dlg_req_within.c:173]: inconsitent dlg timer data on dlg 0xb565c500 [2166:2082643875] with clid ' 5ed68b60-fdbe-4aa7-819e-0002a400e27c(a)64.76.155.86' and tags '5e604a9ea4' 'ed505c1da4' This is the output for dlg_list_ctx kamctl fifo dlg_list_ctx dialog:: hash=2995:55475986 state:: 4 ref_count:: 2 timestart:: 1348177242 timeout:: 15084496 callid:: 33d68b60-a09a-b1dc-8198-0002a400e27c(a)64.76.155.86 from_uri:: sip:557100050994@64.76.155.86 from_tag:: 3360b198a4 caller_contact:: sip:557100050994@64.76.155.86 caller_cseq:: 226 caller_route_set:: caller_bind_addr:: udp:64.76.155.82:5060 callee_bind_addr:: udp:64.76.155.82:5060 to_uri:: sip:0267005622408596@64.76.155.82 to_tag:: ba507815a4 callee_contact:: sip:0267005622408596@64.76.155.130 callee_cseq:: 226 callee_route_set:: context:: I can’t find the problem. I’m using this dialog params : # ---- dialog params ------ modparam("dialog", "dlg_flag", 31) modparam("dialog", "profiles_with_value","gwid" ) modparam("dialog", "default_timeout",7200) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) $avp(i:10)=10 $dlg_ctx(timeout_bye) = 4; dlg_manage(); Regards, Ricardo

2 3

kamailio workers and private memory
by Uri Shacked 25 Sep '12

25 Sep '12

Hi, I use "kamctl sercmd pkg.stats" to check the private memory status. I notice that the workers for the proccess "slow timer", "timer", "MI FIFO", use a lot of memory. My system configuration does "mt_reload" for around 5 million lines every hour, it looks like in time, the pkg memory of the workers above is growing...... I set the private memory to 16M instead of 4M. Now the used size (after 3 weeks) is 4M. I will upgrade to 3.3.0 soon and will use mem_join=1, will it help the private memory defrag as well? How can I increase the number of workers for these processes? Will it help increasing it if possible? BR, Uri

2 1

Issue Starting up Kamalio.
by Jay chary 24 Sep '12

24 Sep '12

Hi We are trying to install Kamailio v3.3.1 on CENTOS, I followed the below urls for installation http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Howeever when I try to start Kamailio by issuing kamctl start , I see the below response : INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed and I see the below in the messages log : Sep 20 10:08:49 kamailio kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Sep 20 10:08:49 kamailio /usr/local/sbin/kamailio[21533]: INFO: usrloc [hslot.c:53]: locks array size 512 Sep 20 10:08:49 kamailio /usr/local/sbin/kamailio[21533]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: auth_db [authdb_mod.c:244]: empty parameter 1 not allowed Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:543 Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:552 Sep 20 10:08:52 kamailio /usr/local/sbin/kamailio[21533]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:585 Sep 20 10:09:30 kamailio avahi-daemon[2760]: Invalid query packet. Can u please help us ? Regards Chary

3 2

Kamailio presentations
by Pavel Segeč 24 Sep '12

24 Sep '12

Hello Everybody, I'm an university teacher and founder of the nil.uniza.sk portal with some SIP/Kamailio tutorials. I'd like to use the Kamailio server as the basement for my class lecturers about the IP multimedia communication systems. May anyone help me and provide a link to usable presentation about the kamailio, which should I use and probably extend in my classes? I search the web but nothing appropriate have not found thanks palo73

1 0

Re: [SR-Users] xcap server support for org.openmobilealliance.xcap-directory
by Peter Dunkley 23 Sep '12

23 Sep '12

Hi, org.openmobilealliance.xcap-directory isn't supported by the xcap-server module. Adding support for this shouldn't be difficult though. Regards, Peter Juha Heinanen <jh(a)tutpro.com> wrote: >in xcap module readme there is an example xcap server config file and it >does not include in xcap-caps aiud >org.openmobilealliance.xcap-directory. does it mean that xcap-directory >application is not currently supported or could the aiud just be added >there? > >-- juha > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 3

xcap server support for org.openmobilealliance.xcap-directory
by Juha Heinanen 22 Sep '12

22 Sep '12

in xcap module readme there is an example xcap server config file and it does not include in xcap-caps aiud org.openmobilealliance.xcap-directory. does it mean that xcap-directory application is not currently supported or could the aiud just be added there? -- juha

1 0

"Command not found" error during "make install"
by Youngjin Park 22 Sep '12

22 Sep '12

Hi, I faced the following error during installing "sip-router" by command "make install" after successful compilation by command "make all". I will appreciate you so much if you can shed light on. My system is 32 bit Fedora. Thanks in advance. Youngjin install -m 644 README /usr/local/share/doc/ser/ docbook2x-man -s ../../doc/stylesheets/serdoc2man.xsl auth.xml make[1]: docbook2x-man: Command not found make[1]: *** [auth.7] Error 127 make: [modules-man] Error 1 (ignored) docbook2x-man -s ../../doc/stylesheets/serdoc2man.xsl auth_db.xml make[1]: docbook2x-man: Command not found make[1]: *** [auth_db.7] Error 127 make: [modules_s-man] Error 1 (ignored) make[1]: `sercmd' is up to date.

3 6

no socket found for match second RR
by Juha Heinanen 21 Sep '12

21 Sep '12

when proxy gets ack to invite, it spits out this kind of error message: Sep 20 22:03:35 wheezy2 /usr/sbin/sip-proxy[4575]: INFO: processing ACK <sip:jh@192.98.103.10:5074> Sep 20 22:03:35 wheezy2 /usr/sbin/sip-proxy[4575]: WARNING: rr [loose.c:744]: no socket found for match second RR route headers in the request are: Route: <sip:192.98.103.10:5060;transport=tcp;r2=on;ftag=43884fd989cff6dc;lr> Route: <sip:192.98.103.10:5060;r2=on;ftag=43884fd989cff6dc;lr> and when i start the proxy it tells this: Listening on udp: 192.98.103.2 [192.98.103.2]:5070 udp: 127.0.0.1:5072 tcp: 192.98.103.2 [192.98.103.2]:5070 tcp: 127.0.0.1:5072 tcp: 127.0.0.1:6060 Aliases: *: 192.98.103.10:5060 *: sip.test.fi:5060 why does loose_route() not find the first alias? -- juha

4 9

any easy way to access record-route URIs?
by Juha Heinanen 21 Sep '12

21 Sep '12

is there any easy way to access URIs in one or more record-route headers? there is @via[index], but no corresponding select for record- route URIs. -- juha

3 5

Re: [SR-Users] [sr-dev] kamailio core at qm_status ( patch required important)
by Jijo 21 Sep '12

21 Sep '12

Hi Daniel, This patch needs to be applied to avoid core. Thanks Jijo On Wed, Sep 19, 2012 at 10:54 AM, Jijo <realjijo(a)gmail.com> wrote: > > Hi All, > > Finally i found the issue, > > Here is one of the bad trace for SUBSCRIBE(722bytes) and NOTIFY(1282bytes) > which corrupted the memory. The messages came in the order NOTIFY and > SUBSCRIBE. The core is generated in a different place but I believe this > could be the reason for memory corruption. > > Here is the trace UDP Process 27294 processing NOTIFY and Process 27303 processing > SUBSCRIBE . > > > > The explanation and implementation is below > > > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27303] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R39 - Entry R-URI=1234 FD=10.233.20.152 > SI=10.233.20.152 > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27294] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R1 - Force LAN socket: tcp:10.233.20.151:5060<null> > > 2012-09-19T02:06:17+01:00 [info] sipserver: [27303] INFO: <script>: > CI=1-3292(a)10.233.20.152 -R1 - Force LAN socket: tcp:10.233.20.151:5060<null> > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:2357]: tcp_conn_send_put : calling wbufq_add > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:730]: ERROR: wbufq_add(722 bytes): buf:SUBSCRIBE > sip:1234@10.233.20.141:5063;transport=tcp SIP/2.0 > > Record-Route: <sip:10.233.20.151;tran > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:747]: ERROR: wbufq_add(722 bytes): first:b00519f4 last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27303] ERROR: <core> > [tcp_main.c:774]: ERROR: wbufq_add(2 last free crt_size:722): > first:b00519f4 last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:796]: ERROR: wbufq_insert(1282 bytes): buf:NOTIFY > sip:1234@10.233.20.141:5063;transport=tcp SIP/2.0 > > Record-Route: <sip:10.233.20.151;transpo > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:801]: ERROR: wbufq_insert(2 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27294] ERROR: <core> > [tcp_main.c:820]: ERROR: wbufq_insert(22 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [err] sipserver: [27359] ERROR: <core> > [tcp_main.c:887]: ERROR: wbufq_run(3 last free ): first:b00519f4 > last:b00519f4 > > 2012-09-19T02:06:17+01:00 [crit] sipserver: [27359] : <core> > [mem/q_malloc.c:157]: BUG: qm_*: fragm. 0xb00519dc (address 0xb00519f4) end > overwritten(0, 0)! > > 2012-09-19T02:06:18+01:00 [alert] sipserver: [27265] ALERT: <core> > [main.c:755]: child process 27359 exited by a signal 11 > > 2012-09-19T02:06:18+01:00 [alert] sipserver: [27265] ALERT: <core> > [main.c:758]: core was generated > > 2012-09-19T02:06:18+01:00 [info] sipserver: [27265] INFO: <core> > [main.c:770]: INFO: terminating due to SIGCHLD > > > > Process 27294(NOTIFY) created the TCP connection structure for destination > IP and just before calling wbufq_insert(), context switch happened and > process 27303(SUBSCRIBE) got the cpu. Since the connection structure is > already available process 27303 add the SUBSCRIBE message(722 bytes) to > wbufq. Afterwards process 27294 got the CPU and invoked wbufq_insert() > which basically corrupted the memory due to an overflow with the existing > implementation. > > > > > > inline static int _wbufq_insert(struct tcp_connection* c, const char* > data, > > > unsigned int size) > > { > > struct tcp_wbuffer_queue* q; > > struct tcp_wbuffer* wb; > > > > q=&c->wbuf_q; > > if (likely(q->first==0)) /* if empty, use wbufq_add */ > > return _wbufq_add(c, data, size); > > : > > : > > : > > : > > if ((q->first==q->last) && > ((q->last->b_size-q->last_used)>=size)){ > > /* one block with enough space in it for > size bytes */ > > > > memmove(q->first->buf+size, > q->first->buf, size); > > memcpy(q->first->buf, data, size); > > q->last_used+=size; > > } > > > > The above condition shall be true in this case and memmove was moving the > pointer which was causing the overflow. > > memmove(void *dest, const void *src, size_t n); > > > > As per the memmove man page, the src shall be copied with size ‘n’ to a > temporary buffer and then temporary buffer to dest. > > > > dest is q->first->buf+size: which is basically (q->first->buf + NOTIFY > MSG SIZE). so the dst will move my 1282 bytes, so we have remaining space > of only 818 bytes. > > > > src is q->first->buf: which is basically copied to temp buffer with NOTIFY > SIZE(1282bytes). > > > > Finally we are moving the buffer from temporary buffer of size 1282 bytes > to buffer which we left with 818 bytes, This basically corrupt the memory > and on wbufq_run we see the memory corruption > > > > 2012-09-19T02:06:17+01:00 [crit] sipserver: [27359] : <core> > [mem/q_malloc.c:157]: BUG: qm_*: fragm. 0xb00519dc (address 0xb00519f4) end > overwritten(0, 0)! > > > > I think we don’t need memove, so we can change the code as below > > > > if ((q->first==q->last) && ((q->last->b_size-q->last_used)>=size)){ > > /* one block with enough space in it for size bytes */ > > //memmove(q->first->buf+size, q->first->buf, size); > > memcpy(q->first->buf+q->last_used, data, size); > > q->last_used+=size; > > } > > OR > we need only the else part as it always add the block to the first. > > Thanks > Jijo > > On Wed, Jul 18, 2012 at 12:42 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> sorry, I just read the last messages in the thread and I didn't noticed >> it is about a crash, but thought it is about an annoying log message. >> >> The backtrace is no longer matching the latest sources of the 3.1 branch, >> but I expect it is due to a double free issue, so you have to update to >> latest 3.1.6, as suggested by other users here. There were many fixes from >> 3.1.0 to 3.1.6. >> >> Cheers, >> Daniel >> >> >> On 7/17/12 6:03 PM, Jijo wrote: >> >> Hi, >> >> This is not happening at shutdown or status check. Its aborting when >> the system is active. >> >> Thanks >> Jijo >> On Tue, Jul 17, 2012 at 3:06 AM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> does it keep being or it is one time and that's it? >>> >>> That is printed at shut down or if pkg_status() or shm_status() is >>> executed from some part of code or in config via cfgutils module functions. >>> >>> You can get rid of them by setting memdbg and memlog to a value higher >>> than debug global parameter. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 7/16/12 5:28 PM, Jijo wrote: >>> >>> Thanks.. It is not easy to upgrade as it is happening at customer >>> system. >>> Is there any change occurred for this issue.I looked at it, but didn't >>> see anything in q_malloc.c/qm_status() >>> >>> >>> On Mon, Jul 16, 2012 at 11:12 AM, Jon Bonilla <manwe(a)aholab.ehu.es>wrote: >>> >>>> El Mon, 16 Jul 2012 10:27:42 -0400 >>>> Jijo <realjijo(a)gmail.com> escribió: >>>> >>>> > Hi All, >>>> > >>>> > I'm observing a core intermittently at "qm_status (qm=0x786cd000) at >>>> > mem/q_malloc.c:763" for kamailio version 3.1.0 >>>> > >>>> >>>> I'd say that you're using a very old version. You should update your >>>> branch to >>>> 3.1.6 or upgrade to a newer branch. >>>> >>>> >>>> _______________________________________________ >>>> sr-dev mailing list >>>> sr-dev(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>> >>> >>> >>> _______________________________________________ >>> sr-dev mailing listsr-dev@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu >>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw >> >> >

2 1

Kamailio OS Sysctl Tweaks
by Cenk İlker İzanlı 20 Sep '12

20 Sep '12

Hi, Is there any sysctl tweaks for OS to get the most of Kamailio as performance meaning? Thx

2 2

Kamailio Presence module
by Gary Shergill 20 Sep '12

20 Sep '12

Hi Kamailio Community, Recently installed Kamailio, plan to integrate with SylkServer. Tried the SIP IM feature, but can't seem to get it working. Logged on to Bria and Blink, two seperate accounts. They can call each other, but can't add each other for presence. So I installed the presence module with apt-get and edited the kamailio.cfg file to add the line; #!define WITH_PRESENCE Restarting the server comes up with the following error; ---------------------- root@opensep:~# vi /etc/kamailio/kamailio.cfg root@opensep:~# service kamailio restart Not starting Kamailio: invalid configuration file! -e 0(5114) : <core> [cfg.y:3591]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 32-36: syntax error 0(5114) : <core> [cfg.y:3591]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 32-36: Invalid arguments 0(5114) : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 392, column 37: ERROR: bad config file (3 errors) ---------------------- Those lines are as follows; ---------------------- #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ---------------------- The exact place is where it says DBURL on both params. I am using Ubuntu 12.04 and Kamailio 3.3.x. Thank you for your time. Kind Regards, Gary Shergill

3 8

authentication for client applications
by David Thomson 20 Sep '12

20 Sep '12

Hi, I am working on a project where a custom sip client will be integrated into a suite of applications to provide voip. The sip client will be working with Kamailio. The goal is to ensure that the client is authorized for communication with kamailio before allowing any calls to be made. Conventional username/password authentication for individual users will also be used once the client has been authenticated. Currently other applications in the suite use a digital signature in the http headers when communicating with server processes. If the signature is validated by the server process then the applications identity is validated and communication with the server process is allowed. Is it possible to include a public key and digital signature in the register events and have kamailio perform the transformation to validate the client's identity? If so which module provides such functionality? Has something like this been implemented in the past? Thanks for any input. ttyl,Dave

3 2

call forking using dbaliases not working for un-NATed clients
by Yufei Tao 19 Sep '12

19 Sep '12

Hi I have a strange problem on forking calls to a group of users. For example I have two users y2 and y3 in dbaliases, both with alias_username 'group'. And y2 and y3 both registers with Kamailio fine. When I make a call to 'group' from a third client y1, what my kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to BRANCH_ALIASDB, where a lookup location will be done for each of the username resulting from lookup of dbaliases, something like this: y1--INVITE 'group'-->lookup dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup location---¬ | |-->relay ---->[BRANCH_ALIASDB]--->'y3'-->lookup location---- The all works well as long as all clients are NAT'ed. However when they are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only goes to one of the group members, e.g. y2 only. When checking the log, it seemed to have done the dbaliases lookup fine, and each location lookup successfully. But Kamailio only relayed y2's IP, e.g. to the client, while y3's to itself. When comparing the location table when clients are NAT'ed or not, I find that the 'received' column is only populated when I do fix_nated_register. And group calls only works when 'received' column is populated. That explains why when clients are NAT'ed group calls work, as I only do fix_nated_register if nat_uac_test returns true. But if this is the only reason, if two clients register using the same username, e.g. both as y3, and when 'received' column of location table is empty (no fix_nated_register done), I would expect a call to y3 should also only make 1 client ring. But in fact both of them rang! The flow is like: y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client registered as 'y3' | ---> IP of 2nd client registered as 'y3' While a call to 'group' (thus dbaliases lookup took place) under such un-NAT'ed set up made only 1 client ring. So I can make it work by always doing fix_nated_register. But I'm not clear about these things: - why does a lookup of dbaliases before lookup of location make such difference? - does lookup location work differently depending on whether it is called from trunk or from a route called from a branch route? Following is relevant parts from my config file: ############################################################# route[LOCATION] { if ( alias_db_lookup("dbaliases") ) { t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB], # call another route that looks up location, # if not existent, call drop() } else { xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n"); route(LOCATION_TRUNK); # normal look up location and sending of 404 etc } ... ... } ############################################################# branch_route[BRANCH_ALIASDB] { xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n"); route(LOCATION_BRANCH); } route[LOCATION_BRANCH] { if (!lookup("location")) { # Drop this branch - it's going nowhere drop(); } } ############################################################# route[RELAY] { xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); 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 (!t_relay()) { sl_reply_error(); } exit; } ############################################################ route[NAT] { #!ifdef WITH_NAT xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); force_rport(); if (nat_uac_test("2")) { if (method=="REGISTER") { fix_nated_register(); xlog("L_DBG","NAT: Just done fix_nated_register in REGISTER message in NAT route\n"); } else { xlog("L_DBG","NAT: fix_nated_contact\n"); fix_nated_contact(); } setflag(FLT_NATS); } # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force media relay #!endif return; } ############################################################# And in the main route, route LOCATION and RELAY are the last two routes: route { ... route(NAT); ... ... # user location service route(LOCATION); route(RELAY); } Hope I have made it clear. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Re: [SR-Users] call forking using dbaliases not working for un-NATed clients
by Yufei Tao 19 Sep '12

19 Sep '12

In LOCATION_BRANCH, after lookup location, I've added: if ($du==$null) $du=$ru; And now everything works! In summary, either of the following would fix the problem: 1. call fix_nated_contact for all clients, so that 'received' column in location table is populated (so that $du will be set), or 2. set $du after lookup if it is empty Wonder if it is worth adding fixes in the source code. Thanks very much Klaus for pointing me to the right direction! Yufei Date: Wed, 19 Sep 2012 15:13:29 +0100 From: Yufei Tao <yufei.tao(a)redembedded.com> Subject: Re: [SR-Users] call forking using dbaliases not working for un-NATed clients To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5059D309.6030500(a)redembedded.com> Content-Type: text/plain; charset="ISO-8859-1" I should have made it clearer: $du is null both before and after lookup location in LOCATION_BRANCH when no fix_nated_register was done (thus 'received' column in location table was null). When fix_nated_register was done, $du for each branch was null before lookup location, but set to 'received' after lookup location as you said. But in this case (fix_nated_register done), everything works fine. So seems now the problem happens when: 'received' in location table is null, causing $du not to be set. I used to think $du is only set by record-route and thanks for clarifying this Even when without fix_nated_register (thus no 'received'), the lookup location for each branch, i.e. y2 and y3, set $ru to the 'contact' field successfully, but $du isn't set. But Kamailio did relay to the first branch (trunk?) and not the others. So the first branch is handled differently? So maybe when $du is null, should set it to $ru on the branches? Yufei On 19/09/12 11:53, Klaus Darilion wrote: > On 19.09.2012 12:24, Yufei Tao wrote: >> Hi Klaus >> >> Thanks for the reply! >> >> I check the $du, it is always null before and after the lookup. Is it >> only set when relaying to a proxy (from record-route), and not to a >> client? > That's strange. For NATed clients, $du must contain the 'received' > URI. Otherwise they can not be contacted as $ru contains the private > IP address. > >> When no fix_nated_register is called, the lookup location for both >> clients y2 and y3 is successful from the log, when printing out $ru >> after lookup. But seems Kamailio only relays to one client's IP while >> not others. I think there must be some differences when branch route is >> executed first time and second time as you said. As it feels like for >> the first branch (trunk?) it used the 'contact' column from the location >> table, and for the other branches, it tries to use 'received'? > It seems db_lookup() creates multiple branches. Is lookup() only > finding 1 contact in table or multiple contacts? > > regards > Klaus > >> Yufei >> >> On 18/09/12 18:49, Klaus Darilion wrote: >>> I suspect that the branch route is first executed for the NATed >>> client. Then the 'received' column is used as destination URI. When >>> executing the branch route again, the destination URI is still the >>> value from the previous branch, and lookup() will not overwrite is as >>> 'received' is not available. Then Kamailio sends the INVITE again to >>> the first client. >>> >>> You can try to set $du to Null before lookup(). ($du=null or >>> $du=$null, not sure what the correct syntax is). >>> >>> Another workaround is to use fix_nated_register() for every client >>> (the pragmatic and more secure approach). >>> >>> regards >>> Klaus >>> >>> On 18.09.2012 15:16, Yufei Tao wrote: >>>> Hi >>>> >>>> I have a strange problem on forking calls to a group of users. For >>>> example I have two users y2 and y3 in dbaliases, both with >>>> alias_username 'group'. And y2 and y3 both registers with Kamailio >>>> fine. >>>> When I make a call to 'group' from a third client y1, what my >>>> kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to >>>> BRANCH_ALIASDB, where a lookup location will be done for each of the >>>> username resulting from lookup of dbaliases, something like this: >>>> >>>> y1--INVITE 'group'-->lookup >>>> dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup >>>> location---? >>>> >>>> | |-->relay >>>> >>>> ---->[BRANCH_ALIASDB]--->'y3'-->lookup >>>> location---- >>>> >>>> The all works well as long as all clients are NAT'ed. However when >>>> they >>>> are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only >>>> goes to one of the group members, e.g. y2 only. When checking the log, >>>> it seemed to have done the dbaliases lookup fine, and each location >>>> lookup successfully. But Kamailio only relayed y2's IP, e.g. to the >>>> client, while y3's to itself. >>>> >>>> When comparing the location table when clients are NAT'ed or not, I >>>> find >>>> that the 'received' column is only populated when I do >>>> fix_nated_register. And group calls only works when 'received' >>>> column is >>>> populated. That explains why when clients are NAT'ed group calls work, >>>> as I only do fix_nated_register if nat_uac_test returns true. >>>> >>>> But if this is the only reason, if two clients register using the same >>>> username, e.g. both as y3, and when 'received' column of location >>>> table >>>> is empty (no fix_nated_register done), I would expect a call to y3 >>>> should also only make 1 client ring. But in fact both of them rang! >>>> The >>>> flow is like: >>>> >>>> y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client >>>> registered as 'y3' >>>> | >>>> ---> IP of 2nd client >>>> registered as 'y3' >>>> >>>> While a call to 'group' (thus dbaliases lookup took place) under such >>>> un-NAT'ed set up made only 1 client ring. >>>> >>>> So I can make it work by always doing fix_nated_register. But I'm not >>>> clear about these things: >>>> - why does a lookup of dbaliases before lookup of location make such >>>> difference? >>>> - does lookup location work differently depending on whether it is >>>> called from trunk or from a route called from a branch route? >>>> >>>> >>>> Following is relevant parts from my config file: >>>> >>>> ############################################################# >>>> route[LOCATION] >>>> { >>>> if ( alias_db_lookup("dbaliases") ) >>>> { >>>> t_on_branch("BRANCH_ALIASDB"); # in >>>> branch_route[BRANCH_ALIASDB], >>>> # call another route that >>>> looks up >>>> location, >>>> # if not existent, call drop() >>>> >>>> } >>>> else >>>> { >>>> xlog("L_DBG","LOCATION: not alias - go to lookup location >>>> trunk\n"); >>>> route(LOCATION_TRUNK); # normal look up location and sending of >>>> 404 etc >>>> } >>>> >>>> ... ... >>>> } >>>> ############################################################# >>>> branch_route[BRANCH_ALIASDB] >>>> { >>>> xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; >>>> Method:$rm\n"); >>>> route(LOCATION_BRANCH); >>>> } >>>> >>>> route[LOCATION_BRANCH] >>>> { >>>> if (!lookup("location")) >>>> { >>>> # Drop this branch - it's going nowhere >>>> drop(); >>>> } >>>> } >>>> ############################################################# >>>> route[RELAY] { >>>> xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); >>>> >>>> >>>> #!ifdef WITH_NAT >>>> if (check_route_param("nat=yes")) { >>>> setbflag(FLB_NATB); >>>> } >>>> if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { >>>> xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); >>>> 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 (!t_relay()) { >>>> sl_reply_error(); >>>> } >>>> >>>> exit; >>>> } >>>> >>>> ############################################################ >>>> route[NAT] { >>>> #!ifdef WITH_NAT >>>> xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); >>>> >>>> force_rport(); >>>> if (nat_uac_test("2")) { >>>> if (method=="REGISTER") { >>>> fix_nated_register(); >>>> xlog("L_DBG","NAT: Just done fix_nated_register in >>>> REGISTER >>>> message in NAT route\n"); >>>> } else { >>>> xlog("L_DBG","NAT: fix_nated_contact\n"); >>>> fix_nated_contact(); >>>> } >>>> setflag(FLT_NATS); >>>> } >>>> >>>> # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force >>>> media relay >>>> #!endif >>>> return; >>>> } >>>> >>>> ############################################################# >>>> And in the main route, route LOCATION and RELAY are the last two >>>> routes: >>>> route { >>>> ... >>>> route(NAT); >>>> >>>> ... ... >>>> >>>> # user location service >>>> route(LOCATION); >>>> >>>> route(RELAY); >>>> } >>>> >>>> Hope I have made it clear. Thanks very much! >>>> >>>> Yufei >>>> -- >>>> Yufei Tao >>>> Red Embedded -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

1 0

IM on Kamailio
by Gary Shergill 19 Sep '12

19 Sep '12

Hi Kamailio Community, I've been configuring the presence module on Kamailio and trying to get it working, using the following as a guideline; http://nil.uniza.sk/instant-messaging/simple/configuring-im-and-presence-ka… >From that I assumed that enabling Presence would mean that IM would be enabled too. However, that doesn't seem to be the case. May I ask please how to configure IM on Kamailio? Note that I am testing this with one computer connected by Bria and another computer connected via Blink. I am able to log on to a user on each (test1 and test2) and they are able to call each other. The issue is, with presence enabled, they are unable to IM each other (or add each other as contacts and see online status). Thank you. Kind Regards, Gary Shergill

4 4

Kamailio-Redirect based LCR routing
by Fatima Chahrour~Vanrise Support 19 Sep '12

19 Sep '12

Dear Daniel\List, Following up to a previous post where I was able in my testing lab to redirect call with 1st destination of lcr_gw table to another Kamailio(acting as any SBC) with Daniels assist, I need a hand in redirecting the call with multiple contacts , basically based on the lcr routing rule configured in kamailio LCR module for the rerouted call. Is this possible using code 300 with multiple choices message? Or you advise me with another method? If 300 redirect message works, what should be the variable set as $ru to get all lcr_gw destinations? Example for clarification: My current redirect script used after lcr functions is : .. $ru= $ru; sl_send_reply("302","Moved Temporary"); .. This script is successfully sending the 302 message with Contact: sip:1119613454646@192.168.111.195. (1119613454646 is the dialed number with prefix, 192.168.111.195 is first lcr_gw ip). Now if the call to 192.168.111.195 is failed and I need to redirect the second destination, second lcr_gw ip, how can I achieve this? Thank you for your usual response. Kind Regards, F Chahrour

2 4

AUTOREPLY call forking using dbaliases not working for...
by strobel＠hfo-telecom.de 18 Sep '12

18 Sep '12

******************* AUTOREPLY VON yufei.tao(a)redembedded.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 17.09.2012 bis einschließlich 21.09.2012 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Künzel oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hi I have a strange problem on forking calls to a group of users. For example I have two users y2 and y3 in dbaliases, both with alias_username 'group'. And y2 and y3 both registers with Kamailio fine. When I make a call to 'group' from a third client y1, what my kamailio.cfg does is: do an alias_db_lookup("dbaliases"), and goes to BRANCH_ALIASDB, where a lookup location will be done for each of the username resulting from lookup of dbaliases, something like this: y1--INVITE 'group'-->lookup dbaliase-->[BRANCH_ALIASDB]--->'y2'-->lookup location---¬ | |-->relay ---->[BRANCH_ALIASDB]--->'y3'-->lookup location---- The all works well as long as all clients are NAT'ed. However when they are not NAT'ed, e.g. all on the same LAN with Kamailio, the call only goes to one of the group members, e.g. y2 only. When checking the log, it seemed to have done the dbaliases lookup fine, and each location lookup successfully. But Kamailio only relayed y2's IP, e.g. to the client, while y3's to itself. When comparing the location table when clients are NAT'ed or not, I find that the 'received' column is only populated when I do fix_nated_register. And group calls only works when 'received' column is populated. That explains why when clients are NAT'ed group calls work, as I only do fix_nated_register if nat_uac_test returns true. But if this is the only reason, if two clients register using the same username, e.g. both as y3, and when 'received' column of location table is empty (no fix_nated_register done), I would expect a call to y3 should also only make 1 client ring. But in fact both of them rang! The flow is like: y1--INVITE 'y3'-->lookup location for 'y3'----> IP of 1st client registered as 'y3' | ---> IP of 2nd client registered as 'y3' While a call to 'group' (thus dbaliases lookup took place) under such un-NAT'ed set up made only 1 client ring. So I can make it work by always doing fix_nated_register. But I'm not clear about these things: - why does a lookup of dbaliases before lookup of location make such difference? - does lookup location work differently depending on whether it is called from trunk or from a route called from a branch route? Following is relevant parts from my config file: ############################################################# route[LOCATION] { if ( alias_db_lookup("dbaliases") ) { t_on_branch("BRANCH_ALIASDB"); # in branch_route[BRANCH_ALIASDB], # call another route that looks up location, # if not existent, call drop() } else { xlog("L_DBG","LOCATION: not alias - go to lookup location trunk\n"); route(LOCATION_TRUNK); # normal look up location and sending of 404 etc } ... ... } ############################################################# branch_route[BRANCH_ALIASDB] { xlog("L_DBG", "BRANCH_ALIASDB: $fU@$fd -> $rU@$rd; Method:$rm\n"); route(LOCATION_BRANCH); } route[LOCATION_BRANCH] { if (!lookup("location")) { # Drop this branch - it's going nowhere drop(); } } ############################################################# route[RELAY] { xlog("L_DBG","RELAY: method=$rm, callid=$ci, cseq=$cs\n"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { xlog("L_DBG", "RELAY: about to call RTPPROXY\n"); 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 (!t_relay()) { sl_reply_error(); } exit; } ############################################################ route[NAT] { #!ifdef WITH_NAT xlog("L_DBG","NAT: method=$rm, callid=$ci, cseq=$cs\n"); force_rport(); if (nat_uac_test("2")) { if (method=="REGISTER") { fix_nated_register(); xlog("L_DBG","NAT: Just done fix_nated_register in REGISTER message in NAT route\n"); } else { xlog("L_DBG","NAT: fix_nated_contact\n"); fix_nated_contact(); } setflag(FLT_NATS); } # setflag(FLT_NATS); ## -- YT: set NAT flag for all, so will force media relay #!endif return; } ############################################################# And in the main route, route LOCATION and RELAY are the last two routes: route { ... route(NAT); ... ... # user location service route(LOCATION); route(RELAY); } Hope I have made it clear. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

record_route_advertised_address question
by Juha Heinanen 18 Sep '12

18 Sep '12

rr readme on record_route_advertised_address function tells: Note: A second Record-Route will be inserted if the transport used on the inbound and outbound interfaces changes. that is not the case. i made call record_route_advertised_address("192.98.103.10:5060"); when request came over udp and went out over tcp and only one rr header was inserted: incoming: U 2012/09/17 17:30:17.467079 192.98.103.10:5060 -> 192.98.103.2:5070 INVITE sip:test@test.fi SIP/2.0. ... outgoing: T 2012/09/17 17:30:17.504631 192.98.103.10:5060 -> 192.98.103.10:38078 [AP] INVITE sip:0x20b7580@192.98.103.10:5050;transport=tcp SIP/2.0. Record-Route: <sip:192.98.103.10:5060;ftag=wumod;lr>. ... i have not set enable_double_rr param and default is 1. is this a bug that i should report on the tracker? -- juha

3 5

Port change 5060
by Vijay Thakur 18 Sep '12

18 Sep '12

All Experts, I want to change my SIP port from 5060 for other one. Before making any change in my live server (Kamailio 3.2.1, i want to be sure. Kindly suggest me that where should i make changes to implement it. This is a security measure for kamailio from port scanning. With thanks in advance. Vijay Thakur

5 6

pidf-lo extraction
by Cenk İlker İzanlı 17 Sep '12

17 Sep '12

Hi, Is there any basic document showing how to extract info from pidf-lo data on kamailio part? Thx.

2 1

rsyslog settings for good kamailio performance
by Albert Petit 17 Sep '12

17 Sep '12

Hi, For load testing I would like to setup rsyslog so Kamailio logs are written asynchronously Would the following configuration be enough for that purpose in * rsyslog.conf*? # # Write log file in asynchronous mode and use an IO Buffer of 64 Kb # IO buffer config http://www.rsyslog.com/doc/rsyslog_conf_global.html # # $OMFileAsyncWriting on $OMFileFlushOnTXEnd off $OMFileFlushInterval 60 $OMFileIOBufferSize 128k #---------------------------------------------------------------------- ## kamailio ##---------------------------------------------------------------------- $outchannel kamailio_log_rotation,/var/log/pocserverplatform/kamailio.log,524288000,/usr/local/bin/kamailio_log_rotation_script :programname, contains, "kamailio" $kamailio_log_rotation local0.* $kamailio_log_rotation Apart of that is there a way to completely disable completely any logs of Kamailio to discard they're causing any performance issue? Thanks Albert

3 2

kamailio prepaid services
by Cenk İlker İzanlı 15 Sep '12

15 Sep '12

Hi, Is there any guide to install/configure kamailio as prepaid service infrastructure? Thx

1 0

Re: [SR-Users] Limiting concurrent calls
by Alex Balashov 15 Sep '12

15 Sep '12

Htable is another popular method for those disposed less favourably to the dialog module. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Uri Shacked <ushacked(a)gmail.com> wrote:in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

1 0

Limiting concurrent calls
by Uri Shacked 15 Sep '12

15 Sep '12

in a few words - you can set avp's as the profile params in the dialog module (that module works great for me) that way you wont need to set more than one profile. and for every call use it. or just use $shv() and +1 or -1 every INVITE, CANCEL, BYE or failure reply. hope it helps, Uri > Hi, > I need to limit the number of concurrent calls go through kamailio. > Its a simple set up(Not per user basis, a global limit will suffice). > > I know i need to employ dialog module. But i'm a little confused about > profiles. I thought for simple application like this, i may be able to > use profile altogether. > > Any code snippet to limit number of concurrent calls will be very helpful. > > Thanks in advance. > -- > -aft

1 0

I get no calls?
by gooly 15 Sep '12

15 Sep '12

Hi, I have registered my (Fritz!Box) ATA successfully registered on my account 8305234(a)iptl.org but what is tried I never received a call? Here is what I can see on 'my' web-page with you: > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:59 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:58 500 Server Internal Error > "C.A. Schreiber" <sip:17772184364@callcentric.com> non-local 2012-09-14 14:56 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:45 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-09-10 13:43 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:40 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "alles" <sip:192.168.0.201:5070@188.26.245.223:5060> non-local 2012-09-10 13:39 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:20 500 Server Internal Error > "C.A. Schreiber" <sip:8305234@iptel.org> on line 2012-08-22 00:19 500 Server Internal Error Please can you tell me what I have to do to be able to receive Voip-Calls? Regards, Carl

2 1

Limiting concurrent calls
by Aft nix 15 Sep '12

15 Sep '12

Hi, I need to limit the number of concurrent calls go through kamailio. Its a simple set up(Not per user basis, a global limit will suffice). I know i need to employ dialog module. But i'm a little confused about profiles. I thought for simple application like this, i may be able to use profile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft

1 0

cfg. parser: failed to find command force_rtp_proxy
by Cenk İlker İzanlı 15 Sep '12

15 Sep '12

Hi, I'm installing kamailio and rtpproxy with this guide. http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour Versions of the programs: # aptitude versions rtpproxy i 1.2.1-1 stable 500 root@li304-234:~# kamailio -V version: kamailio 3.3.0 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 22:18:19 Jun 18 2012 with gcc 4.4.5 When I start kamailio, I get this error, Sep 14 17:08:44 (none) kamailio: ERROR: <core> [cfg.y:3455]: cfg. parser: failed to find command force_rtp_proxy Sep 14 17:08:44 (none) kamailio: : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command, missing loadmodule?#012 Sep 14 17:08:44 (none) kamailio: ERROR: <core> [cfg.y:3455]: cfg. parser: failed to find command force_rtp_proxy Sep 14 17:08:44 (none) kamailio: : <core> [cfg.y:3594]: parse error in config file /etc/kamailio/kamailio.cfg, line 806, column 19: unknown command, missing loadmodule?#012 Related lines throwing error are below. 724 route[RTPPROXY] { 725 #!ifdef WITH_NAT 726 if (is_method("BYE")) { 727 unforce_rtp_proxy(); 728 } else if (is_method("INVITE")){ 729 force_rtp_proxy(); 730 } 731 if (!has_totag()) add_rr_param(";nat=yes"); 732 #!endif 733 return; 734 } 735 803 #!ifdef WITH_NAT 804 if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) 805 && status=~"(183)|(2[0-9][0-9])") { 806 force_rtp_proxy(); 807 } 808 if (isbflagset("6")) { 809 fix_nated_contact(); 810 } 811 #!endif Does anyone know what's the problem with cfg file? Thx. Cenk

2 2

[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Joe Hart 15 Sep '12

15 Sep '12

Hi all, For a project on which I'm currently working, I am having some problems figuring out how to correctly configure Kamailio to communicate with RTP Proxy in order to send media into and out of a network with private IP address ranges. I have a proxy set up to send the SIP traffic, and all of this is working fine. However, I'm having some trouble getting the RTP Proxy set up. Currently, when the call is connected, the offer/answer is made and RTP Proxy seems to be taking over, but I'm having trouble getting my audio to flow in both directions. Examination of the traffic coming into and out of this machine seems to indicate that the IP addresses aren't being mangled correctly. Specifically, it appears the internal IP address isn't being changed to reflect the IP address of the machine on which RTP Proxy is running, so that when the caller tries to send audio back, the IP it's given to reply to is 10.10.x.x, which obviously won't work. I have tried experimenting with specifically setting IP addresses in the rtpproxy_offer() and _answer() methods to no avail, as well as setting various flags in those methods. However, I must admit that I'm not entirely sure what's happening under the hood with these methods, or what rtpproxy is doing with that information when it gets it. Rather than continue to hack at this by trial and error, I'm hoping someone here can point me in the right direction. Any advice, example code or pep talks would be greatly appreciated. Thanks in advance, -- Joe Hart Voice Systems Integrator Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0673

8 37

Howto add peers and routing rules
by Joel Smith | VOZELIA 15 Sep '12

15 Sep '12

Hi, I'm getting started with kamailio and siremis. I've just finished getting the server ready and the software installed. Once in the siremis admin menu, i've added a domian and a subscriber and I have my softphone registered with kamailio. My doubt is, where do I add peers so I can set up the routing rules? Are there any tutorials? Thanks in advanced. Best regards. -- Joel Smith Cell: +34 639 03 13 53 E-Mail: joel(a)vozelia.com <joel(a)vozelia.com> <http://www.vozelia.com> <http://twitter.com/vozelia> <http://www.facebook.com/pages/Vozelia-Operador-de-telefonia-IP-para-empresa…>

2 1

Re: [SR-Users] using kamailio with clients in a nat environment
by David Thomson 15 Sep '12

15 Sep '12

Sorry, his name is Daniel Goepp. -----Original Message----- From: jdavidthomson(a)hotmail.com Date: Sat, 15 Sep 2012 03:53:51 To: Daniel-Constantin Mierla<miconda(a)gmail.com> Reply-To: jdavidthomson(a)hotmail.com Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment I got it working. The advertise parameter wasn't available in the initial versions of rtpproxy I was using. A guy named daniel geopp has a patched version that enables advertisement of the public ec2 ip. Thanks for your help :) Ttyl, Dave -----Original Message----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Tue, 11 Sep 2012 06:24:33 To: <jdavidthomson(a)hotmail.com> Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment Hello, the sdp does not show that rtpproxy was engaged. Check your config, you can use debugger module with cfgtrace on to see what actions are executed. Also, probably you have to advertise the public ip address of your ec2 instance -- see second parameter for rtpproxy module functions. Cheers, Daniel On 9/11/12 3:52 AM, David Thomson wrote: Hi, I'm using rtpproxy and per the documentation: rtpproxy -l public_ip -s udp:localhost:22222 -F Attached is the following: Dave registering Daniel registering Dave calling Daniel, where Dave has a public IP and Daniel is behind a nat. Please let me know what you think is up. ttyl, Dave [...] # U 207.219.69.217:40821 -> 10.248.96.110:5060 INVITE sip:daniel@54.245.31.65 SIP/2.0. Via: SIP/2.0/UDP 10.207.158.89:51362;rport;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 70. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Route: <sip:54.245.31.65;transport=udp;lr> <sip:54.245.31.65;transport=udp;lr> . Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. # U 10.248.96.110:5060 -> 207.219.69.217:40821 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.207.158.89:51362;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF;received=207.219.69.217. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Server: kamailio (3.3.0 (x86_64/linux)). Content-Length: 0. . # U 10.248.96.110:5060 -> 75.119.228.57:5060 INVITE sip:daniel@192.168.1.102:5060;transport=udp;registering_acc=54_245_31_65 SIP/2.0. Record-Route: <sip:54.245.31.65;lr=on;nat=yes> <sip:54.245.31.65;lr=on;nat=yes> . Via: SIP/2.0/UDP 54.245.31.65:5060;branch=z9hG4bKd6f6.b6db86e5.0. Via: SIP/2.0/UDP 10.207.158.89:51362;received=207.219.69.217;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 69. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Re: [SR-Users] using kamailio with clients in a nat environment
by David Thomson 15 Sep '12

15 Sep '12

I got it working. The advertise parameter wasn't available in the initial versions of rtpproxy I was using. A guy named daniel geopp has a patched version that enables advertisement of the public ec2 ip. Thanks for your help :) Ttyl, Dave -----Original Message----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Tue, 11 Sep 2012 06:24:33 To: <jdavidthomson(a)hotmail.com> Cc: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] using kamailio with clients in a nat environment Hello, the sdp does not show that rtpproxy was engaged. Check your config, you can use debugger module with cfgtrace on to see what actions are executed. Also, probably you have to advertise the public ip address of your ec2 instance -- see second parameter for rtpproxy module functions. Cheers, Daniel On 9/11/12 3:52 AM, David Thomson wrote: Hi, I'm using rtpproxy and per the documentation: rtpproxy -l public_ip -s udp:localhost:22222 -F Attached is the following: Dave registering Daniel registering Dave calling Daniel, where Dave has a public IP and Daniel is behind a nat. Please let me know what you think is up. ttyl, Dave [...] # U 207.219.69.217:40821 -> 10.248.96.110:5060 INVITE sip:daniel@54.245.31.65 SIP/2.0. Via: SIP/2.0/UDP 10.207.158.89:51362;rport;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 70. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Route: <sip:54.245.31.65;transport=udp;lr> <sip:54.245.31.65;transport=udp;lr> . Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. # U 10.248.96.110:5060 -> 207.219.69.217:40821 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.207.158.89:51362;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF;received=207.219.69.217. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Server: kamailio (3.3.0 (x86_64/linux)). Content-Length: 0. . # U 10.248.96.110:5060 -> 75.119.228.57:5060 INVITE sip:daniel@192.168.1.102:5060;transport=udp;registering_acc=54_245_31_65 SIP/2.0. Record-Route: <sip:54.245.31.65;lr=on;nat=yes> <sip:54.245.31.65;lr=on;nat=yes> . Via: SIP/2.0/UDP 54.245.31.65:5060;branch=z9hG4bKd6f6.b6db86e5.0. Via: SIP/2.0/UDP 10.207.158.89:51362;received=207.219.69.217;rport=40821;branch=z9hG4bKPj7ohi7tDsVbjxTKXgNQsTviH-zPxv0rBF. Max-Forwards: 69. From: <sip:dave@54.245.31.65> <sip:dave@54.245.31.65> ;tag=BDvpvvb0IsNu9oFPCeO2XI-M-Qwl7V4S. To: <sip:daniel@54.245.31.65> <sip:daniel@54.245.31.65> . Contact: <sip:dave@207.219.69.217:40821;ob> <sip:dave@207.219.69.217:40821;ob> . Call-ID: cG7lF7dXytDwgmhVa19g0aGEOa6s1Rxl. CSeq: 17407 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_SGH-T989D-15/r1841. Content-Type: application/sdp. Content-Length: 425. . v=0. o=- 3556317015 3556317015 IN IP4 10.207.158.89. s=pjmedia. t=0 0. m=audio 4008 RTP/AVP 96 3 0 8 101. c=IN IP4 10.207.158.89. a=rtcp:4009 IN IP4 10.207.158.89. a=sendrecv. a=rtpmap:96 SILK/8000. a=fmtp:96 useinbandfec=0. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=zrtp-hash:1.10 0a851ee8921d1f71658c8253dd6097893c48e40886759ec0e21e79a61c1f1289. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

avp_radius load time high
by Ricardo Martinez 14 Sep '12

14 Sep '12

> Hello List. > > I’m having the next issue. I have a kamailio v3.2 working with a Radius > server as a backend. All the INVITE’s coming to the kamailio server are > challenged against the Radius server and then avp_load is called from > Radius server too. I’m having excessive delay times when I have near 10 > call per second. So I find it very weird. I have made a benchmark for the > avp_load : > > bm_start_timer("timer-tranum"); > > if( !radius_load_caller_avps("$fU@$fd")) { > > sl_send_reply("403","Forbidden - Estamos experimentando problemas"); > > exit; > > } else { > > xlog("L_INFO", "[$ci]:[AUTH_REQUEST]: ok"); > > }; > > bm_log_timer("timer-tranum"); > > > > and this is the output : > > > > Sep 12 18:26:07 benchmark (timer timer-tranum [1]): 112441 [ > msgs/total/min/max/avg - LR: > 100/18446744073695545751/23297/18446744073709236247/184467440736955456.000000 > | GB: > 2100/18446744073708471097/10746/18446744073709542058/8784163844623081.000000] > > Sep 12 18:26:55 benchmark (timer timer-tranum [1]): 280543 [ > msgs/total/min/max/avg - LR: > 100/10238598/21317/18446744073708596212/102385.980000 | GB: > 2200/9158079/10746/18446744073709542058/4162.763182] > > Sep 12 18:28:20 benchmark (timer timer-tranum [1]): 312902 [ > msgs/total/min/max/avg - LR: > 100/5048656/15491/18446744073709546022/50486.560000 | GB: > 2300/14206735/10746/18446744073709546022/6176.841304] > > Sep 12 18:28:31 benchmark (timer timer-tranum [1]): 301818 [ > msgs/total/min/max/avg - LR: > 100/5622716/43785/18446744073708810522/56227.160000 | GB: > 2400/19829451/10746/18446744073709546022/8262.271250] > > Sep 12 18:28:44 benchmark (timer timer-tranum [1]): 18446744073708689124 > [ msgs/total/min/max/avg - LR: > 100/1329199/59661/18446744073709167366/13291.990000 | GB: > 2500/21158650/10746/18446744073709546022/8463.460000] > > Sep 12 18:29:26 benchmark (timer timer-tranum [1]): 44160 [ > msgs/total/min/max/avg - LR: > 100/18446744073701320457/19576/18446744073708922050/184467440737013216.000000 > | GB: 2600/12927491/10746/18446744073709546022/4972.111923] > > Sep 12 18:30:18 benchmark (timer timer-tranum [1]): 218573 [ > msgs/total/min/max/avg - LR: > 100/18446744073707306935/17459/18446744073708876520/184467440737073056.000000 > | GB: 2700/10682810/10746/18446744073709546022/3956.596296] > > Sep 12 18:31:06 benchmark (timer timer-tranum [1]): 132751 [ > msgs/total/min/max/avg - LR: > 100/9611955/17057/18446744073708726431/96119.550000 | GB: > 2800/20294765/10746/18446744073709546022/7248.130357] > > Sep 12 18:31:55 benchmark (timer timer-tranum [1]): 18859 [ > msgs/total/min/max/avg - LR: > 100/18446744073705270953/18859/18446744073708747515/184467440737052704.000000 > | GB: 2900/16014102/10746/18446744073709546022/5522.104138] > > Sep 12 18:32:48 benchmark (timer timer-tranum [1]): 61954 [ > msgs/total/min/max/avg - LR: > 100/7124234/18036/18446744073708596768/71242.340000 | GB: > 3000/23138336/10746/18446744073709546022/7712.778667] > > Sep 12 18:33:39 benchmark (timer timer-tranum [1]): 53489 [ > msgs/total/min/max/avg - LR: > 100/18446744073707180114/18177/18446744073708790040/184467440737071808.000000 > | GB: 3100/20766834/10746/18446744073709546022/6698.978710] > > Sep 12 18:34:35 benchmark (timer timer-tranum [1]): 60604 [ > msgs/total/min/max/avg - LR: > 100/18446744073701108277/33983/18446744073708839351/184467440737011072.000000 > | GB: 3200/12323495/10746/18446744073709546022/3851.092187] > > Sep 12 18:35:24 benchmark (timer timer-tranum [1]): 54705 [ > msgs/total/min/max/avg - LR: > 100/7185862/15880/18446744073708608536/71858.620000 | GB: > 3300/19509357/10746/18446744073709546022/5911.926364] > > Sep 12 18:36:21 benchmark (timer timer-tranum [1]): 60356 [ > msgs/total/min/max/avg - LR: > 100/18446744073705699677/19466/18446744073708844524/184467440737056992.000000 > | GB: 3400/15657418/10746/18446744073709546022/4605.122941] > > Sep 12 18:37:14 benchmark (timer timer-tranum [1]): 18446744073708679441 > [ msgs/total/min/max/avg - LR: > 100/3506212/16102/18446744073708791471/35062.120000 | GB: > 3500/19163630/10746/18446744073709546022/5475.322857] > > Sep 12 18:38:07 benchmark (timer timer-tranum [1]): 93933 [ > msgs/total/min/max/avg - LR: > 100/18446744073704680760/17569/18446744073708741284/184467440737046816.000000 > | GB: 3600/14292774/10746/18446744073709546022/3970.215000] > > Sep 12 18:38:58 benchmark (timer timer-tranum [1]): 142777 [ > msgs/total/min/max/avg - LR: > 100/681953/15933/18446744073708708694/6819.530000 | GB: > 3700/14974727/10746/18446744073709546022/4047.223514] > > Sep 12 18:39:44 benchmark (timer timer-tranum [1]): 165921 [ > msgs/total/min/max/avg - LR: > 100/5153736/15749/18446744073708657622/51537.360000 | GB: > 3800/20128463/10746/18446744073709546022/5296.963947] > > Sep 12 18:40:31 benchmark (timer timer-tranum [1]): 127615 [ > msgs/total/min/max/avg - LR: > 100/18446744073709161716/20690/18446744073708793559/184467440737091616.000000 > | GB: 3900/19738563/10746/18446744073709546022/5061.170000] > > Sep 12 18:41:21 benchmark (timer timer-tranum [1]): 47762 [ > msgs/total/min/max/avg - LR: > 100/9001175/18909/18446744073708632234/90011.750000 | GB: > 4000/28739738/10746/18446744073709546022/7184.934500] > > Sep 12 18:42:11 benchmark (timer timer-tranum [1]): 62944 [ > msgs/total/min/max/avg - LR: > 100/18446744073708163674/17934/18446744073708825788/184467440737081632.000000 > | GB: 4100/27351796/10746/18446744073709546022/6671.169756] > > > > > > If the output is in microseconds and I have from time to time (for the > last 100 messages) values like 184467440737081632.000000 microsecs > > IS THIS NORMAL? > > Then it backs to a more rational value : 90011.750000 microsecs. > > I also made measures in my Radius Server and is answering with average > time around 0.01 secs. > > So… what could it be the problem?.. Could be the RadiusClient-ng ? > > In my attempt to solve the issue I increased the childrens for the > kamailio server from 16 to 60, but the problem persists. > > Can someone help me here? > > > > Best Regards, > > Ricardo Martinez.- > > > > > > >

3 7

collecting stats per subscriber
by Krishna Kurapati 14 Sep '12

14 Sep '12

Hi, Is there a module that can be used to collect number of MESSAGEs sent or received by a subscriber; number of registrations by a subscriber? These will be per subscriber as oppose to statistics collected for entire system. Thanks Krish Kura

2 2

How to do RTPProxy load balancing?
by Mino Haluz 14 Sep '12

14 Sep '12

Hi, mediaproxy has dispatcher and relay, so it load balances the traffic automatically. How it is done with rtpproxy ? Thanks, Mino

3 4

FAQ page on the wiki
by Daniel-Constantin Mierla 14 Sep '12

14 Sep '12

Hello, I just started a FAQ page on the wiki site. Hopefully people will collect there useful content, based on their experience, questions that repeat on the mailing lists or other sources. Added just a few content for the moment, to show the syntax -- the page is available at: * http://www.kamailio.org/wiki/tutorials/faq/main It is using the QNA plugin for dokuwiki: * https://www.dokuwiki.org/plugin:qna Anyone can make an own account to edit the wiki, the registration form is available at: * https://www.kamailio.org/wiki/start?do=register Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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Re: [SR-Users] Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

thank you for helps cenk and pavel. I see that I had wrong information about kamailio.

2 1

Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

juha writes: > this kind of questions have been asked many times on this list. search > archive for responses. you are right maybe but it is hard to search the archieve. And if it is easier to say "serach" than please show me way to search the arcieve quickly. In this link I m trying to search archieve month by month: http://lists.sip-router.org/pipermail/sr-users/ I m new and thanks your kind help...

3 2

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 14 Sep '12

14 Sep '12

Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or CANCEL. It'll just figure out what to do on its own. None of this has to do with dialog state, though. Just rtpproxy control. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: > >> Peter: Thanks for the tip! Really interesting. But I do not >> understand, why also this list contains the calls that were ended by >> sipp... Should I search for some mistake in my kamaillio config ? > > Perhaps you don't close them with unforce_rtp_proxy: > > if(method=="BYE" || method=="CANCEL"){ > unforce_rtp_proxy(); > } > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > 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

4 7

Re: [SR-Users] Max-Forwards: and loops. Testing purpose.
by Daniel-Constantin Mierla 14 Sep '12

14 Sep '12

Hello, do not write private emails from mailing list discussions, if not sensitive data was asked explicitly. If there is no rule to breaking the loop, after some time the message gets very big. Cheers, Daniel On 9/14/12 7:03 AM, Anton Kvashenkin wrote: > Thanks for respond. I just have used forward() based on RURI. > > got > > 0(2558) ERROR: <core> [msg_translator.c:1835]: ERROR: > build_req_buf_from_sip_req: out of memory > 0(2558) ERROR: <core> [forward.c:601]: ERROR: forward_request: > building failed > 0(2558) WARNING: <core> [receive.c:211]: WARNING: receive_msg: error > while trying script > > With maxfwd it works like a charm. http://paste.ubuntu.com/1204093/ > > 2012/9/12 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > > On 9/11/12 1:12 PM, Anton Kvashenkin wrote: > > Hi, List. > > I'm using sipsak to fire some predefined sip message with > Max-Forwards: header equal 0 to test mf_process_maxfwd_header > function. I'm doing this exclusively for learning purpose. As > far as we know, this exported function from maxfwd module > services only for adding Max-Forwards header, decrement or > detect Max-Forwards: 0 header to prevent loops. So, my > question guys, how can I create this loop in my lab, with two > proxies for example? What message should I craft? > > what kind of message are you looking to craft? > > Looping from one proxy to another can be achieved with > t_relay("_the other proxy address_") function from tm module. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - > http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - > http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - > http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

Does kamailio supports g729 codec? Or which codecs does kamailio supports? Where or how to learn the supported or unsupported codecs for kamailio? Thanks in advance...

2 1

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

That's a rather dated study. But it's better than a nonexistent reference point, true. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:According to this (http://transnexus.com/index.php/performance-test-results-for-openser-and-rt…) "For a server hosting both OpenSER and RTPproxy, each 1 GHz of CPU processing capacity can manage a maximum of 325 simultaneous calls." I have 2.4GHz for rtpproxy, but CPU/Mem/network is ok, so the bottleneck should be somewhere else probably.. On Thu, Sep 13, 2012 at 5:56 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > I'm not sure what a single instance of rtpproxy can handle, but most people > squeezing thousand of concurrent calls per box are probably doing it on > multicore boxes by binding multiple instances of rtpproxy with different > core affinities, and round-robining among them. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/ > > Mino Haluz <mino.haluz(a)gmail.com> wrote: > The results: > > - rtpproxy calls count 280 > - sipp calls count 2000 > - iptraf on proxy 4.8MB/s > - G711a codec > > So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), > rtpproxy calls count is really the right value. CPU usage is ok on > every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy > cannot serve more than 270-280 calls ? > > On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: >> Ok, so I put there unforce_rtp_proxy even though I'm using >> rtpproxy_manage. The tip with nc now really shows the calls count. >> >> But the dialog count is still higher and higher, so I have bug >> somewhere in the configuration. I'll check it. >> >> On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov >> <abalashov(a)evaristesys.com> wrote: >>> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE >>> or >>> CANCEL. It'll just figure out what to do on its own. >>> >>> None of this has to do with dialog state, though. Just rtpproxy control. >>> >>> >>> >>> >>> -- Alex >>> >>> -- >>> Sent from my Samsung mobile, and thus lacking in the refinement one might >>> expect from a proper keyboard. >>> >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 235 E Ponce de Leon Ave >>> Suite 106 >>> Decatur, GA 30030 >>> Tel: +1-678-954-0670 >>> Web: http://www.evaristesys.com/ >>> >>> Mino Haluz <mino.haluz(a)gmail.com> wrote: >>> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >>> >>> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> >>> wrote: >>>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>>> >>>>> Peter: Thanks for the tip! Really interesting. But I do not >>>>> understand, why also this list contains the calls that were ended by >>>>> sipp... Should I search for some mistake in my kamaillio config ? >>>> >>>> Perhaps you don't close them with unforce_rtp_proxy: >>>> >>>> if(method=="BYE" || method=="CANCEL"){ >>>> unforce_rtp_proxy(); >>>> } >>>> >>>> -- >>>> With best regards, Peter Lemenkov. >>>> >>>> _______________________________________________ >>>> 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 >>> >>> _______________________________________________ >>> 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

1 0

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

I'm not sure what a single instance of rtpproxy can handle, but most people squeezing thousand of concurrent calls per box are probably doing it on multicore boxes by binding multiple instances of rtpproxy with different core affinities, and round-robining among them. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:The results: - rtpproxy calls count 280 - sipp calls count 2000 - iptraf on proxy 4.8MB/s - G711a codec So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), rtpproxy calls count is really the right value. CPU usage is ok on every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy cannot serve more than 270-280 calls ? On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: > Ok, so I put there unforce_rtp_proxy even though I'm using > rtpproxy_manage. The tip with nc now really shows the calls count. > > But the dialog count is still higher and higher, so I have bug > somewhere in the configuration. I'll check it. > > On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or >> CANCEL. It'll just figure out what to do on its own. >> >> None of this has to do with dialog state, though. Just rtpproxy control. >> >> >> >> >> -- Alex >> >> -- >> Sent from my Samsung mobile, and thus lacking in the refinement one might >> expect from a proper keyboard. >> >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Decatur, GA 30030 >> Tel: +1-678-954-0670 >> Web: http://www.evaristesys.com/ >> >> Mino Haluz <mino.haluz(a)gmail.com> wrote: >> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >> >> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: >>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>> >>>> Peter: Thanks for the tip! Really interesting. But I do not >>>> understand, why also this list contains the calls that were ended by >>>> sipp... Should I search for some mistake in my kamaillio config ? >>> >>> Perhaps you don't close them with unforce_rtp_proxy: >>> >>> if(method=="BYE" || method=="CANCEL"){ >>> unforce_rtp_proxy(); >>> } >>> >>> -- >>> With best regards, Peter Lemenkov. >>> >>> _______________________________________________ >>> 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 >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>

2 1

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Really? Interesting, I had no idea. I thought the rtpproxy control protocol was binary and did not lend itself easily to interaction in this manner. Thanks for the tip. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > You can't get it from rtpproxy. You'd really have to use something like the > dialog or htable modules to keep call state and get that from Kamailio. On the contrary it's possible (using raw UDP reads/writes): work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 sessions created: 0 active sessions: 0 active streams: 0 work ~: Where * h1u203u03 is randomly chosen token, * 127.0.0.1 is the rtpproxy's control IP, * 22222 is the rtpproxy's control port, * "-u" means that we're using UDP * -w 1 is the timeout in seconds to wait before closing nc. I can't imagine that someone will use nc in performance testing but I think it looks like a good start. -- With best regards, Peter Lemenkov.

3 4

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Thanks for this info. Netcat is a really underappreciated tool. I can't count how many times I've unnecessarily written utilities to send stuff on UDP sockets, forgetting that nc can do it. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. Yep, this command protocol allows us to do tricks like this which is quite helpful for debugging. Unfortunately this particular command ( 'I' and its counterpart 'Ib' ) seems to be broken - it doesn't prepend received random cookie at the beginning of the answer. Just FYI - I collected some examples of control commands here (just grep for ser_proto:decode): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_proto_test.erl and here (which could be even more interesting since it contains an example of a real session - create session, lookup, destroy, get stats, etc): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_test.erl > Thanks for the tip. Anytime :) -- With best regards, Peter Lemenkov.

1 0

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

If the dialog module is working correctly, and the SIP flows are standards-compliant, the dialog module should automatically track all subsequent state changes and remove calls from tracking after a BYE is processed. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Ok, I'm tagging dialogs with dlg_manage(), but even if the call ends, it still keeps info about this dialog in list "kamctl fifo dlg_list". Should I somehow close the dialog when the BYE transaction is ended ? Peter: Thanks for the tip! Really interesting. But I do not understand, why also this list contains the calls that were ended by sipp... Should I search for some mistake in my kamaillio config ? On Thu, Sep 13, 2012 at 3:57 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. > Thanks for the tip. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/ > > Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: >> You can't get it from rtpproxy. You'd really have to use something like >> the >> dialog or htable modules to keep call state and get that from Kamailio. > > On the contrary it's possible (using raw UDP reads/writes): > > work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 > sessions created: 0 > active sessions: 0 > active streams: 0 > work ~: > > Where > > * h1u203u03 is randomly chosen token, > * 127.0.0.1 is the rtpproxy's control IP, > * 22222 is the rtpproxy's control port, > * "-u" means that we're using UDP > * -w 1 is the timeout in seconds to wait before closing nc. > > I can't imagine that someone will use nc in performance testing but I > think it looks like a good start. > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > 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 >

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

You can't get it from rtpproxy. You'd really have to use something like the dialog or htable modules to keep call state and get that from Kamailio. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Hi, I'm doing the performance test with kamailio + RTPProxy, but I would like to get the real calls count that the rtpproxy is serving. I don't want to use value that I get from sipp. So is there any management tool for rtpproxy, or should I get it somewhere in kamailio config ? Thanks, Mino _______________________________________________ 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

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RTPProxy with kamailio : How to get calls count?
by Mino Haluz 13 Sep '12

13 Sep '12

Hi, I'm doing the performance test with kamailio + RTPProxy, but I would like to get the real calls count that the rtpproxy is serving. I don't want to use value that I get from sipp. So is there any management tool for rtpproxy, or should I get it somewhere in kamailio config ? Thanks, Mino

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script stops processing when relaying e2e ack?
by Andrew Mortensen 12 Sep '12

12 Sep '12

Hi all. I'm working on a custom module and would like to be able to send UPDATEs to the call legs after the script forwards the e2e ACK from the caller. I had hoped to export a function to call in my ACK handling route after the script forwards the ACK. However, the script stops processing after calling t_relay in my ACK handling route. I tracked this back to the fact that t_relay_to invokes forward_request when handling e2e ACKs, and returns the return code of forward_request to the script. Since forward_request returns 0 on success, the t_relay call in the script returns 0, which stops script processing. Is this intentional? My module currently hooks the TMCB_E2EACK_IN, and I can UPDATE the caller here, but the callee seems to be ignoring the UPDATE if it's received prior to the ACK. Thanks for any advice you can offer. andrew

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Kamailio 3.2, NAT, rtpproxy bridging between interfaces
by Russell McConnachie 12 Sep '12

12 Sep '12

Hello, I'm having some issues getting media bridging working in kamailio and rtpproxy. I've attempted to attach as much relevant information as possible. Thanks in advance for taking a look; I'm completely stuck right now. Actual results:No audio is being passed through the rtpproxy from caller "A" to caller "B" even though both media server and UAC are sending media to the rtpproxy. Expected results:Audio that is sent to rtpproxy is bridged/proxied across from caller "A" to caller "B" and vice-versa. ------------------------------------------------------------------------------- Signaling trace: ------------------------------------------------------------------------------- 08:34:57.543224 IP (tos 0x0, ttl 49, id 3237, offset 0, flags [none], proto: UDP (17), length: 1456) x.x.x.199.31280 > x.x.x.102.sip: SIP, length: 1428 INVITE sip:9991@r18.test-bwks.local:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5 Route: <sip:x.x.x.102:5060;lr> Max-Forwards: 70 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "99999999" <sip:99999999@192.168.2.250:51620;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.1007 Content-Type: application/sdp Content-Length: 619 v=0 o=MxSIP 0 0 IN IP4 192.168.2.250 s=SIP Call c=IN IP4 192.168.2.250 t=0 0 m=audio 51720 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv 08:34:57.545766 IP (tos 0x10, ttl 64, id 58136, offset 0, flags [none], proto: UDP (17), length: 439) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 411 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280;received=x.x.x.199 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Server: kamailio (3.2.4 (x86_64/linux)) Content-Length: 0 08:34:57.547619 IP (tos 0x10, ttl 64, id 47936, offset 0, flags [+], proto: UDP (17), length: 1500) 10.0.0.38.sip > 10.0.0.29.sip: SIP, length: 1472 INVITE sip:9991@10.0.0.29 SIP/2.0 Record-Route: <sip:10.0.0.38;r2=on;lr=on;nat=yes> Record-Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes> Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0 Via: SIP/2.0/UDP 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5 Max-Forwards: 69 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "99999999" <sip:99999999@x.x.x.199:31280;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.1007 Content-Type: application/sdp Content-Length: 629 v=0 o=MxSIP 0 0 IN IP4 10.0.0.38 s=SIP Call c=IN IP4 10.0.0.38 t=0 0 m=audio 29682 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event 08:34:57.551126 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 409) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 381 SIP/2.0 100 Trying Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060> Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Content-Length:0 08:34:57.561084 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 723) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 695 SIP/2.0 180 Ringing Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Content-Length:0 08:34:57.562404 IP (tos 0x10, ttl 64, id 58137, offset 0, flags [none], proto: UDP (17), length: 672) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 644 SIP/2.0 180 Ringing Via:SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Content-Length:0 08:34:57.609771 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 953) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 925 SIP/2.0 200 OK Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/media_control+xml,application/sdp,multipart/mixed Content-Type:application/sdp Content-Length:133 v=0 o=BroadWorks 196 1 IN IP4 10.0.0.24 s=- c=IN IP4 10.0.0.24 t=0 0 m=audio 10498 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 08:34:57.612047 IP (tos 0x10, ttl 64, id 58138, offset 0, flags [none], proto: UDP (17), length: 930) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 902 SIP/2.0 200 OK Via:SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/media_control+xml,application/sdp,multipart/mixed Content-Type:application/sdp Content-Length:161 v=0 o=BroadWorks 196 1 IN IP4 x.x.x.102 s=- c=IN IP4 x.x.x.102 t=0 0 m=audio 29774 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sdpmangled:yes 08:34:57.844645 IP (tos 0x0, ttl 49, id 3239, offset 0, flags [none], proto: UDP (17), length: 506) x.x.x.199.31280 > x.x.x.102.sip: SIP, length: 478 ACK sip:10.0.0.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9 Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes>, <sip:10.0.0.38;r2=on;lr=on;nat=yes> Max-Forwards: 70 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID: b2af8b83e20b0740 CSeq: 29663 ACK User-Agent: Aastra 57i/2.6.0.1007 Content-Length: 0 08:34:57.845182 IP (tos 0x10, ttl 64, id 47937, offset 0, flags [none], proto: UDP (17), length: 509) 10.0.0.38.sip > 10.0.0.29.sip: SIP, length: 481 ACK sip:10.0.0.29:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9 Max-Forwards: 69 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID: b2af8b83e20b0740 CSeq: 29663 ACK User-Agent: Aastra 57i/2.6.0.1007 Content-Length: 0 ------------------------------------------------------------------------------- The call session is up at this point. CPE Router, Source NAT IP x.x.x.199, Internal Network: 192.168.2.0/24 192.168.2.250 is an Aastra 6757i IP Phone, behind the x.x.x.102 is the external interface on our SIP proxy allowing traffic on: Inbound: port 5060/udp Inbound: range 16384 - 32768/udp 10.0.0.38 is the internal interface facing our BroadWorks lab cluster 10.0.0.29 is the internal interface of our BroadWorks Application Server 10.0.0.24 is the internal interface of our BroadWorks Media Server RTP Proxy has been spawned with the following command line arguments: /usr/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy \ -u root \ -s udp:localhost:7722 \ -l x.x.x.102/10.0.0.38 \ -d DBUG LOG_LOCAL5 \ -m 16384 -M 32768 We've changed the /etc/kamailio/kamailio.cfg and added the #!define's at the top of the configuration file: #!define WITH_NAT #!define WITH_DEBUG In the route[NATMANAGE] { } block we've added a few additional lines to handle the setup of the rtpproxy_manage() function which handles the rtpproxy_offer(), rtpproxy_answer() and unforce_rtp_proxy() convientenly in one function. The logic we've got in place currently is: if ((src_ip =~ "10\.0\.0\.*") && (dst_ip =~ "10\.0\.0\.*")) { xlog("L_INFO", "Media session for ($ci) with source address ($si): Internal > External: SDP contact/origin will be x.x.x.102"); rtpproxy_manage("FAIEOC", "x.x.x.102"); } else if (!(src_ip =~ "10\.0\.0\.*") && (dst_ip == x.x.x.102)) { xlog("L_INFO", "Media session for ($ci) with source address ($si): External > Internal: SDP contact/origin will be 10.0.0.38"); rtpproxy_manage("FAEIOC", "10.0.0.38"); } else { xlog("L_INFO", "!!! Uncaught condition for media session for ($ci) with source address ($si) !!!"); } This rtpproxy log is not from the call above, but the same source #, destination # were used: Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new session 84bf2a7cb9d4ab24, tag dcfa6a860e;1 requested, type strong Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new session on a port 28282 created, tag dcfa6a860e;1 Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: pre-filling caller's address with 192.168.2.250:51720 Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: lookup on ports 28282/22500, session timer restarted Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: pre-filling callee's address with 10.0.0.24:10502 ... Call is currently active and then hung-up ... Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:process_rtp: session timeout Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: session on ports 28282/22500 is cleaned up Example of the RTP: 08:53:13.555834 IP (tos 0x0, ttl 48, id 510, offset 0, flags [none], proto: UDP (17), length: 200) x.x.x.199.32052 > x.x.x.102.16552: UDP, length 172 x.x.x.199:32052 > x.x.x.102:16552 Why is this packet not being forwarded to the media proxy (rtpproxy) to 10.0.0.24 (specified in the 200 OK from the BroadWorks Application Server) 08:53:13.562206 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 200) 10.0.0.24.10504 > 10.0.0.38.17968: UDP, length 172 10.0.0.24:10504 > 10.0.0.38:17968 Why is this packet not being forwarded from rtpproxy to the UAC at x.x.x.199? ------------------------------------------------------------------------------- System details: ------------------------------------------------------------------------------- OS: CentOS 5.8 x86-64, Hypervisor is qemu-kvm Memory: 2GB Disk: 40G VCPU: 1 /etc/sysctl.conf: # Controls IP packet forwarding net.ipv4.ip_forward = 1 ------------------------------------------------------------------------------- Current software versions: ------------------------------------------------------------------------------- [root@sip-proxy1 kamailio]# rpm -qa | grep -E 'kamailio|rtpproxy' | xargs rpm -qi Name : rtpproxy Relocations: (not relocatable) Version : 1.2.1 Vendor: Fedora Project Release : 2.el5 Build Date: Sun 19 Sep 2010 08:12:23 AM EDT Install Date: Mon 10 Sep 2012 09:23:28 PM EDT Build Host: x86-10.phx2.fedoraproject.org Group : Applications/Internet Source RPM: rtpproxy-1.2.1-2.el5.src.rpm Size : 105682 License: BSD Signature : DSA/SHA1, Mon 20 Sep 2010 12:40:46 PM EDT, Key ID 119cc036217521f6 Packager : Fedora Project URL : http://www.rtpproxy.org Summary : A symmetric RTP proxy Description : This is symmetric RTP proxy designed to be used in conjunction with the SIP Express Router (SER) or any other SIP proxy capable of rewriting SDP bodies in SIP messages that it processes. Name : kamailio Relocations: (not relocatable) Version : 3.2.4 Vendor: kamailio.org Release : 1.1 Build Date: Fri 03 Aug 2012 05:14:14 AM EDT Install Date: Tue 11 Sep 2012 08:22:32 AM EDT Build Host: build08 Group : Productivity/Telephony/SIP/Servers Source RPM: kamailio-3.2.4-1.1.src.rpm Size : 10200784 License: GPL Signature : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID c9a75909941fdbdd Packager : Daniel-Constantin Mierla <miconda(a)gmail.com> URL : http://kamailio.org/ Summary : Kamailio, very fast, reliable and flexible SIP Server Description : Kamailio is a very fast, reliable and flexible SIP (RFC3261) proxy server. Written entirely in C, kamailio can handle thousands calls per second even on low-budget hardware. A C Shell like scripting language provides full control over the server's behaviour. It's modular architecture allows only required functionality to be loaded. Among available features: IPv4, IPv6, digest authentication, accounting, CPL scripts, instant messaging, MySQL and UNIXODBC support, SIMPLE presence agent, radius authentication, record routing, SMS gateway, ENUM, UDP, TCP, TLS and SCTP, transaction and dialog module, OSP module, statistics support, registrar and user location, SNMP, SIMPLE Presence and Perl programming interface. Name : kamailio-cpl Relocations: (not relocatable) Version : 3.2.4 Vendor: kamailio.org Release : 1.1 Build Date: Fri 03 Aug 2012 05:14:14 AM EDT Install Date: Tue 11 Sep 2012 08:22:33 AM EDT Build Host: build08 Group : Productivity/Telephony/SIP/Servers Source RPM: kamailio-3.2.4-1.1.src.rpm Size : 296947 License: GPL Signature : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID c9a75909941fdbdd Packager : Daniel-Constantin Mierla <miconda(a)gmail.com> URL : http://kamailio.org/ Summary : CPL module (CPL interpreter engine) for Kamailio Description : The kamailio-cpl package provides a CPL interpreter engine for Kamailio

2 1

Max-Forwards: and loops. Testing purpose.
by Anton Kvashenkin 12 Sep '12

12 Sep '12

Hi, List. I'm using sipsak to fire some predefined sip message with Max-Forwards: header equal 0 to test mf_process_maxfwd_header function. I'm doing this exclusively for learning purpose. As far as we know, this exported function from maxfwd module services only for adding Max-Forwards header, decrement or detect Max-Forwards: 0 header to prevent loops. So, my question guys, how can I create this loop in my lab, with two proxies for example? What message should I craft?

2 1

using kamailio with clients in a nat environment
by David Thomson 12 Sep '12

12 Sep '12

hi, I was wondering if in clients in a nat environment need to be in the dmz or need to have ports forwarded for kamailio to be used to connect the clients? I have setup kamailio with the nat settings but am still only getting one way audio. chances are it has not been setup correctly but just checking. If i try to connect the two clients from the same network i can get zrtp and 2 way communication working correctly. if one of the clients is on a different network the call initiates and answers but zrtp and 2way audio does not work. kamailio 3.3 running on ec2 ttyl,Dave

2 4

Re: [SR-Users] Jurisdictional (inter vs intra calling) LCR on Kamailio
by Alex Balashov 11 Sep '12

11 Sep '12

William, This is a business-layer function that is far outside the scope of what built-in Kamailio modules can do. In fact, even the 'lcr' module is something of a misnomer: it does not do LCR in a truly financialised sense, to say nothing of jurisdictionally specific regulatory awareness. It's just a basic weighted routing table. The appropriate place to find such functionality, unless you want to write it yourself, is in a variety of ITSP-oriented commercial platforms built on top of Kamailio. They exist precisely to fill these gaps and bring the somewhat general capabilities of Kamailio into an applied dimension that is useful in specific business and product contexts. Many members of this list represent software or consulting companies that offer such solutions. However, as it is a noncommercial list, I'll refrain from plugging them directly. You may want to ask the question on Kamailio-Business. You can also email me privately and I would be happy to give you some unbiased pointers about the commercial ecosystem. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/William Fulton <wfulton(a)thirdhatch.com> wrote:I can’t find any information anywhere on how to get jurisdictional lcr configured. I have intra and interstate rates for each carrier rate deck and don’t know how to let the system know when one is appropriate versus the other. Can someone point me to a tutorial or some documentation on this? Thanks so much! William

1 0

Jurisdictional (inter vs intra calling) LCR on Kamailio
by William Fulton 11 Sep '12

11 Sep '12

I can't find any information anywhere on how to get jurisdictional lcr configured. I have intra and interstate rates for each carrier rate deck and don't know how to let the system know when one is appropriate versus the other. Can someone point me to a tutorial or some documentation on this? Thanks so much! William

1 0

rls notify delay
by Juha Heinanen 11 Sep '12

11 Sep '12

one rls thing that stilll wonders me is the about 2-5 sec delay that occurs when backend notifies have been received and notify to subscriber ua is sent: Sep 11 17:27:41 siika /usr/sbin/sip-proxy[4495]: INFO: Routing PUBLISH <sip:foo@test.fi> to presence server Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: Handling PUBLISH <sip:foo@test.fi> Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: Routing locally generated NOTIFY <sip:rls@127.0.0.1:5082;transport=tcp> to <sip:jh@test.fi> from <sip:foo@test.fi> Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: presence [notify.c:1575]: NOTIFY sip:jh@test.fi via sip:rls@127.0.0.1:5082;transport=tcp on behalf of sip:foo@test.fi for event presence Sep 11 17:27:41 siika /usr/sbin/pres-serv[19271]: INFO: Handling in-dialog NOTIFY <sip:rls@127.0.0.1:5082;transport=tcp> from <sip:foo@test.fi> to <sip:jh@test.fi> Sep 11 17:27:46 siika /usr/sbin/pres-serv[19265]: INFO: Routing locally generated NOTIFY <sip:cyvrphbz@192.98.103.1:60122;alias=192.98.103.1~47419~2;transport=tcp> to <sip:jh@test.fi> from <sip:jh-buddies@test.fi> Sep 11 17:27:46 siika /usr/sbin/sip-proxy[4482]: INFO: Routing in-dialog NOTIFY <sip:cyvrphbz@192.98.103.1:60122;transport=tcp> from <sip:jh-buddies@test.fi> to <sip:192.98.103.1:47419;transport=tcp> this does not depend on the pua patch nor which db_mode i have in rls module. -- juha

2 2

rls does not generally work
by Juha Heinanen 11 Sep '12

11 Sep '12

it turned out that presence rls module does not work unless pua module is configured to use database mode and database is postgre. the issue is serious, since besides not working, it makes presence/rls server not to accept any new requests. since there is no easy cure, should i go and write a warning in rls readme file so that others don't stumble on this? -- juha

2 6

siptrace and homer
by Krishna Kurapati 11 Sep '12

11 Sep '12

Hi, We have been using homer for some time. The logging is stopping once the disk is full, which make sense. However, it would nice if either kamailio or homer has an option to auto-truncate the siptrace database tables once they grow large. Is there such a feature? Regards. Krishn Kura

2 1

Продукция из Украины
by andrei＠iptel.org 11 Sep '12

11 Sep '12

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: ðÒÉÏÂÒÅÔÅÎÉÅ, ÔÁÍÏÖÅÎÎÏÅ ÏÆÏÒÍÌÅÎÉÅ É ÄÏÓÔÁ×ËÁ ÌÀÂÏÊ ÉÎÔÅÒÅÓÕÀÝÅÊ ÷ÁÓ ÐÒÏÄÕËÃÉÉ ÉÚ õËÒÁÉÎÙ. éÍÅÅÍ ÍÎÏÇÏÌÅÔÎÉÊ ÏÐÙÔ ÜËÓÐÏÒÔÎÙÈ ÏÐÅÒÁÃÉÊ. ôÅÒÒÉÔÏÒÉÁÌØÎÏ ÎÁÈÏÄÉÍÓÑ × ÓÁÍÏÍ ÃÅÎÔÒÅ õËÒÁÉÎÙ. éÍÅÅÍ ×ÓÅ ÎÅÏÂÈÏÄÉÍÙÅ ÜËÓÐÏÒÔÎÙÅ ÒÁÚÒÅÛÅÎÉÑ É ÌÉÃÅÎÚÉÉ. ó 1998 ÇÏÄÁ Ñ×ÌÑÅÍÓÑ ÄÅÊÓÔ×ÕÀÝÉÍÉ ÞÌÅÎÁÍÉ ôÏÒÇÏ×Ï ðÒÏÍÙÛÌÅÎÎÏÊ ðÁÌÁÔÙ õËÒÁÉÎÙ. [...] Content analysis details: (16.9 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.0 FSL_HELO_NON_FQDN_1 FSL_HELO_NON_FQDN_1 3.8 HELO_LOCALHOST HELO_LOCALHOST 2.8 FUZZY_XPILL BODY: Attempt to obfuscate words in spam 3.0 BAYES_95 BODY: Bayes spam probability is 95 to 99% [score: 0.9893] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [123.27.115.159 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?123.27.115.159>] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [123.27.115.159 listed in bb.barracudacentral.org] 0.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [123.27.115.159 listed in dnsbl.sorbs.net] 0.8 RCVD_IN_SORBS_WEB RBL: SORBS: sender is an abusable web server 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines

1 0

Re: [SR-Users] MTREE reload question
by Daniel-Constantin Mierla 11 Sep '12

11 Sep '12

Hello, the used size is pretty much the same after the reload, so doesn't look as leak. If you do 2-3 reloads is shmem:used_size staying around same value? It is clear boost in the fragments, I would say that after start the number of fragments is quite low, because it should contain the good records in memory as well. Btw, the version 3.3.x has options for defragmentation, see the core cookbook, you can enable it. Cheers, Daniel On 9/9/12 10:18 AM, Uri Shacked wrote: > Hi, > here is the statistics after kamailio starts: > shmem:fragments = 28 > shmem:free_size = 3800871312 > shmem:max_used_size = 494132368 > shmem:real_used_size = 494095984 > shmem:total_size = 4294967296 > shmem:used_size = 342642072 > here it is after reload of the num table: > shmem:fragments = 9161531 > shmem:free_size = 3654274496 > shmem:max_used_size = 959885552 > shmem:real_used_size = 640692800 > shmem:total_size = 4294967296 > shmem:used_size = 342654552 > here is the cfg part for mtree: > #------- mtree params ------------- > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=odr;dbtable=service_odr_view;type=0;") > modparam("mtree", "mtree", "name=oper;dbtable=service_oper_type;type=0;") > modparam("mtree", "mtree", > "name=permis;dbtable=service_permisions_to_oper;type=0;") > modparam("mtree", "mtree", > "name=num;dbtable=service_numbers_to_areas_view;type=0;") > modparam("mtree", "char_list", "0123456789") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > here is the number of raws from the DB: > SELECT count(*) FROM `service`.`service_numbers_to_areas_view` = 4195528 > and attached is the memory log. > thanks, > Uri > > On Fri, Sep 7, 2012 at 10:26 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > is this taken only after startup? Get one at startup and another > one after reload, so it can be compared. > > Cheers, > Daniel > > > On 9/6/12 9:22 AM, Uri Shacked wrote: >> Hi, >> here: >> shmem:fragments = 143898 >> shmem:freesize = 3446570952 >> shmem:max_used_size = 861854768 >> shmem:real_used_size = 848396344 >> shmem:total_size = 4294967296 >> shmem:used_size = 319676976 >> >> >> On Thu, Sep 6, 2012 at 9:49 AM, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> >> On 9/5/12 3:06 PM, Uri Shacked wrote: >> >> Hi, >> I use MTREE to load 5 million rows from the database. it >> takes about 30 sec to start kamailio and it is running great. >> Whern I check the shmem usage I see the data take about >> 0.8G out of 4G i set on shmem. >> When i reload the data while kamailio is running, the >> memory usage rises to 1.3G and stays there (the second >> and next reloads stays on 1.3G as well). >> Why doesn't it return to 0.8G after the reload is completed? >> >> >> can you send the statistics related to shmem? >> >> kamctl fifo get_statistics shmem: >> >> There are different values there, some usage come from >> overhead of memory chunks management. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda >> <http://twitter.com/#%21/miconda> - >> http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - >> http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >> http://asipto.com/u/katu >> >> > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu > > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 8

Memory allocation
by David 11 Sep '12

11 Sep '12

Hello, I originally asked this on IRC, but realized that this is probably better suited for the mailing list : If I set memory usage to 100mb ( -m 100 ) and set 100 children ( children=100 in kamailio.cfg ). Will Kamailio use 100 mb or 10, 000 mb ( 10 gb ) of memory ? Is the memory limit per child or global for kamailio ? Is there an MI command I can send to check what kamailio's health ? ( load, free memory, etc... ) ? Thanks, David

2 3

Rtpproxy re-INVITE handling
by Spencer Thomason 11 Sep '12

11 Sep '12

Hello, I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in a largely stock script. Everything works great until the far end (on a public ip) sends a t.38 re-INVITE. The 200OK from the NATed UAC then doesn't trigger rtpproxy and the private IP in the sdp causes the fax to fail. Any help handling these re-INVITEs would be greatly appreciated. I'm happy to post traces if that helps describe the situation. The network topology looks like this: NATed UAC -> Kamailio on a public IP -> UAS on a public IP Thanks in advance, Spencer Connected by DROID on Verizon Wireless

2 12

pua 'Could not convert temporary dialog into a dialog' error
by Juha Heinanen 10 Sep '12

10 Sep '12

when testing presence/rls, i'm getting a pua error when presence server handles subscribe: Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: Handling SUBSCRIBE <sip:test@test.fi> Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: Routing locally generated NOTIFY to <sip:rls@127.0.0.1:5082;transport=tcp> Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: presence [notify.c:1581]: NOTIFY sip:jh@test.fi via sip:rls@127.0.0.1:5082;transport=tcp on behalf of sip:test@test.fi for event presence Sep 8 20:36:12 siika /usr/sbin/pres-serv[22931]: INFO: <core> [tcp_main.c:2787]: quick connect for 0x7f7114cfa490 Sep 8 20:36:12 siika /usr/sbin/pres-serv[22933]: ERROR: pua [send_subscribe.c:689]: Could not convert temporary dialog into a dialog what does pua error message 'Could not convert temporary dialog into a dialog' mean? when does it happen? -- juha

2 9

Example required for XMLRPC module.
by SamyGo 10 Sep '12

10 Sep '12

Hello, I recently posted email relating to using MI_XMLRPC for kamailio monitoring. As Daniel suggested and as mentioned in module docs Mi-xmlrpc is getting obsolete and don't compile on newer OSs. So I need to use XMLRPC module for pulling statistics from kamailio like get_statistics all: Please can anyone point me to some help other than module doc. I need a simple example script to pull data from kamailio and print it. Any help or pointers will be greatly appreciated. Thanks, Sammy G.

1 0

event_route[dialog:failed] and Warning "script writer didn't realese transaction"
by Uri Shacked 10 Sep '12

10 Sep '12

Hi, I started using testing the event_route[dialog:failed]. On my scenario, I receive an INVITE, use t_newtran(), check some stuff and send a failure reply like 3XX or 4XX. I use send_reply() and can see that the TM sends the reply. Now, if I use event_route[dialog:failed], I keep on getting the warning "script writer didn't release transaction". If I delete the event_route[dialog:failed] from the script, the warning does not appear. My solution was to work stateless, which is OK for this scenario. But, still, What does this event_route has that shows this warning? Why doesn't the transaction being released? (I tried to use t_reply as well and still the same). Thanks, Uri

2 1

Re: [SR-Users] MTREE reload question
by Uri Shacked 09 Sep '12

09 Sep '12

> > Hi, > > here is the statistics after kamailio starts: > > shmem:fragments = 28 > shmem:free_size = 3800871312 > shmem:max_used_size = 494132368 > shmem:real_used_size = 494095984 > shmem:total_size = 4294967296 > shmem:used_size = 342642072 > > here it is after reload of the num table: > > shmem:fragments = 9161531 > shmem:free_size = 3654274496 > shmem:max_used_size = 959885552 > shmem:real_used_size = 640692800 > shmem:total_size = 4294967296 > shmem:used_size = 342654552 > here is the cfg part for mtree: > > #------- mtree params ------------- > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=odr;dbtable=service_odr_view;type=0;") > modparam("mtree", "mtree", "name=oper;dbtable=service_oper_type;type=0;") > modparam("mtree", "mtree", > "name=permis;dbtable=service_permisions_to_oper;type=0;") > modparam("mtree", "mtree", > "name=num;dbtable=service_numbers_to_areas_view;type=0;") > modparam("mtree", "char_list", "0123456789") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > > here is the number of raws from the DB: > > SELECT count(*) FROM `service`.`service_numbers_to_areas_view` = 4195528 > > i have the memory log in a file if needed > > thanks, > Uri > > > On Fri, Sep 7, 2012 at 10:26 AM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> is this taken only after startup? Get one at startup and another one >> after reload, so it can be compared. >> >> Cheers, >> Daniel >> >> >> On 9/6/12 9:22 AM, Uri Shacked wrote: >> >> Hi, >> >> here: >> >> shmem:fragments = 143898 >> shmem:freesize = 3446570952 >> shmem:max_used_size = 861854768 >> shmem:real_used_size = 848396344 >> shmem:total_size = 4294967296 >> shmem:used_size = 319676976 >> >> >> >> On Thu, Sep 6, 2012 at 9:49 AM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 9/5/12 3:06 PM, Uri Shacked wrote: >>> >>>> Hi, >>>> I use MTREE to load 5 million rows from the database. it takes about 30 >>>> sec to start kamailio and it is running great. >>>> Whern I check the shmem usage I see the data take about 0.8G out of 4G >>>> i set on shmem. >>>> When i reload the data while kamailio is running, the memory usage >>>> rises to 1.3G and stays there (the second and next reloads stays on 1.3G as >>>> well). >>>> Why doesn't it return to 0.8G after the reload is completed? >>>> >>> >>> can you send the statistics related to shmem? >>> >>> kamctl fifo get_statistics shmem: >>> >>> There are different values there, some usage come from overhead of >>> memory chunks management. >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.com >>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - >>> http://asipto.com/u/kat >>> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >>> http://asipto.com/u/katu >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >> >> > >

1 0

Possible bug in authentication
by martian＠centrum.sk 07 Sep '12

07 Sep '12

Hello to everybody. I am currently working with Kamailio 3.3.1 on RedHat. The "loose_route" function was not working correctly and I observed some very strange behaviour (not as one described in the documentation of the function). I have found that there needs to be a port included in the "alias" variable for the loose_route function to work correctly. However, upon adding the port to alias, the INVITE messages were no longer authenticated (Kamailio just accepted them and didn't send proxy-auth header in 407 message). My alias: alias="domain.ch:5060" Examining default routing logic, I found the problem here: if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ... } The "from_uri==myself" was no longer evaluated as true, because there was a port at the end of the alias. The FROM Header of the INVITE messages looks like: From: "acc1" ;tag=12345 ..so .. no port number there. Btw, I have fixed this with replacing the "myself" list with my own defined variable MY_DOMAIN. #!define MY_DOMAIN ".*(a)domain.ch" So now the condition looks like this: if (is_method("REGISTER") || from_uri=~MY_DOMAIN) { ... } I am not sure if this is a bug that needs to be fixed or not. I am just pointing my finger at it and I hope it will contribute to the development. Also, a valid description of this behavior (when using port in alias) would be appreciated. Cheers, Martin

3 8

Wrong index on expires column in pua module
by Juha Heinanen 07 Sep '12

07 Sep '12

looks like unique constraint on expires_idx has not been removed yet, because i still see Duplicate entry for key 'expires_idx' errors. i'll go and make the changes. -- juha

1 0

FW: Kamailio as sipRedirect
by Fatima Chahrour~Vanrise Support 07 Sep '12

07 Sep '12

Dears, I have an installed 3CX where I want to generate a call from it to the Kamailo to get a list of suppliers to terminate the call to. Thus, the Kamailio should act as a sip redirect to the 3CX (for routing decision). Have you tried such configuration before? And how do I configure the 3CX on the Kamailio? Regards, F Charour

3 12

kamctlrc SQL definitions and db/schema/entities.xml
by Vieri 07 Sep '12

07 Sep '12

Hi, What is the purpose of the "SQL definitions" settings found in kamctlrc? It is also stated that one should edit db/schema/entities.xml but I can't seem to find this file. kamctlrc contains: # SQL definitions # If you change this definitions here, then you must change them # in db/schema/entities.xml too. # FIXME # FOREVER="2020-05-28 21:32:15" # DEFAULT_ALIASES_EXPIRES=$FOREVER # DEFAULT_Q="1.0" # DEFAULT_CALLID="Default-Call-ID" # DEFAULT_CSEQ="13" # DEFAULT_LOCATION_EXPIRES=$FOREVER Thanks, Vieri

2 5

siptrace module with Kamailio 3.3.0 and Siremis 3.2
by Chris Telford 07 Sep '12

07 Sep '12

Hi, Were attempting to get siptrace working on Kamailio 3.3.0 with Siremis 3.2... We've enabled the module, and configured the basic parameters, however were having problems with where exactly in Kamailio's config to call siptrace(). We currently have it at the very end of FAILURE_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE as recommended at the end, and it is logging traces sucessfully. The problem were having, is it is only producing traces with inbound sip messages only. It is also duplicating almost every message (so were clearly call siptrace() too many times per request). Is there a guide on this? or any sample config files to help us speed this process up a little? Regards, Chris Telford

1 0

Re: [SR-Users] sr-users Digest, Vol 88, Issue 19
by Vijay Thakur 07 Sep '12

07 Sep '12

On Friday 07 September 2012 02:34 PM, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Monitoring Kamailio using mi_xmlrpc good or bad ? > (Daniel-Constantin Mierla) > 2. Re: Video Call Quality (Tayeb Meftah) > 3. Re: FW: Kamailio as sipRedirect (Fatima Chahrour~Vanrise Support) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 07 Sep 2012 10:36:50 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Monitoring Kamailio using mi_xmlrpc good or > bad ? > To: SamyGo <govoiper(a)gmail.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5049B222.6080405(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > On 9/7/12 10:21 AM, SamyGo wrote: >> Thanks alot for your time and valuable feedback on the topic. I just >> wanted to be sure that when if comes to collecting stats like 150+ values > did you defined custom stats? IIRC, we don't have so many in the code. > >> from Kamailio per 30 seconds will it impact on the actual performance >> of Kamailio as sip server. I will definitely look for XMLRPC only as >> well if thats scalable. > Stats are just counters, long values, access to them should be very > fast. mi_xmlrpc is a dedicated process, not handling sip at all. > > Cheers, > Daniel >> Thanks >> Sammy >> >> >> >> On Fri, Sep 7, 2012 at 12:35 PM, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> if you fetch the statistics, should be ok. >> >> xmlrpc module should be more scalable that mi_xmlrpc, but uses a >> different command format and output. >> >> Cheers, >> Daniel >> >> >> On 9/6/12 3:24 PM, SamyGo wrote: >>> Hello, >>> >>> I've just finished monitoring all the core modules of kamailio >>> via XML-RPC. I use a script placed on kamailio server which sends >>> command to Kamailio over MI_XMLRPC and fetch details of each >>> requested module. The results fetched are then sent back to the >>> monitoring tool for graphing using SNMP OIDs. >>> >>> *My question* here is that if I query kamailio like this for like >>> 7~10 module over xmlrpc per 30/45 seconds how much overhead will >>> it cause to the Kamailio server !? Is this a good approach or bad ? >>> >>> >>> Thanks, >>> Sammy >>> >>> >>> >>> _______________________________________________ >>> 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 >> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu >> >> Thanks, Tayeb for your reply. What can be checked at client end. I am using Kamailio 3.2. With thanks in advance. Vijay Thakur

1 0

Video Call Quality
by Vijay Thakur 07 Sep '12

07 Sep '12

Dear All Experts, Recently, i have configure my first kamailio server. After some technical hurdles, it is working very fine. There is a issue related to video call. When i make a video call from jitsi to jitsi, iphone to iphone or android, the image is blur, choppy. After googling for this issue, i have found no satisfactory answer. Kindly suggest me the reasons for this issue and what action can be taken. Thanks in advance , Vijay Tha

2 1

Monitoring Kamailio using mi_xmlrpc good or bad ?
by SamyGo 07 Sep '12

07 Sep '12

Hello, I've just finished monitoring all the core modules of kamailio via XML-RPC. I use a script placed on kamailio server which sends command to Kamailio over MI_XMLRPC and fetch details of each requested module. The results fetched are then sent back to the monitoring tool for graphing using SNMP OIDs. *My question* here is that if I query kamailio like this for like 7~10 module over xmlrpc per 30/45 seconds how much overhead will it cause to the Kamailio server !? Is this a good approach or bad ? Thanks, Sammy

2 3

[ ... ] operator
by Vitaliy Aleksandrov 07 Sep '12

07 Sep '12

"Migrating kamailio v1.5.x to v.3.0.0" says that "test" operator no longer exists. But it is mentioned in Core Cookbook 3.2 (and 3.3) and can confuse someone. Can somebody remove it from the documentation ?

2 3

MTREE reload question
by Uri Shacked 07 Sep '12

07 Sep '12

Hi, I use MTREE to load 5 million rows from the database. it takes about 30 sec to start kamailio and it is running great. Whern I check the shmem usage I see the data take about 0.8G out of 4G i set on shmem. When i reload the data while kamailio is running, the memory usage rises to 1.3G and stays there (the second and next reloads stays on 1.3G as well). Why doesn't it return to 0.8G after the reload is completed? BR, Uri

2 3

User accounting
by Ali Jawad 06 Sep '12

06 Sep '12

Hi Is it possible to store user logon/logout dates + user agent used. I know that the info is in the location table but it is not persistent, and I am using the accounting table to store calls. But I am not sure there is a way to store logon/logout/user agent. Any input please ? Regards

2 8

Siremis v3.3.0 Released
by Elena-Ramona Modroiu 06 Sep '12

06 Sep '12

Hi, Siremis v3.3.0 was released - the web management interface compatible with Kamailio v3.3.x. This version is an update of v3.2.1 to be able to work with the database schema of Kamailio v3.3.x. The news of the release, including links to installation and screenshots, is available at: - http://siremis.asipto.com/2012/09/06/siremis-v3-3-0-released/ Regards, Ramona

1 0

Meeting in London
by Carsten Bock 06 Sep '12

06 Sep '12

Hi, anyone from the Kamailio-Community in for a beer tomorrow evening in London? I have several meetings during the day (last one at 5PM) and i'll be travelling back to Germany on Saturday, so i'm free for some chat about Kamailio, SIP, IMS or IPv6 during the evening... :-) If yes, just drop me a mail... Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

Информация для бухгалтера
by jan＠iptel.org 06 Sep '12

06 Sep '12

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: òÁÂÏÔÁÅÔÅ ÂÕÈÇÁÌÔÅÒÏÍ? ðÏÍÎÉÔÅ, ËÁË ÓÌÏÖÎÏ ×ÁÍ ÒÁÎØÛÅ ÂÙÌÏ ÎÁÊÔÉ ÏÂÕÞÅÎÉÅ ÐÏ ÎÕÖÎÏÊ ÔÅÍÅ ôÅÐÅÒØ, ÎÁ ÓÁÊÔÅ ÎÁÛÅÇÏ ÕÞÅÂÎÏÇÏ ÃÅÎÔÒÁ ÎÁ ÜÔÏ ÕÊÄÅÔ ×ÓÅÇÏ ÎÅÓËÏÌØËÏ ÍÉÎÕÔ! é ÎÅ ×ÁÖÎÏ, × ÞÅÍ ×ÁÍ ÎÕÖÎÏ ÒÁÚÏÂÒÁÔØÓÑ - × îäó, ÒÁÓÞÅÔÅ ÏÔÐÕÓËÏ×, ÏÐÌÁÔÅ ÔÒÕÄÁ, ÎÁÌÏÇÏÏÂÌÏÖÅÎÉÉ, ÓÔÒÁÈÏ×ÙÈ ×ÚÎÏÓÁÈ, ÐÏÄÇÏÔÏ×ÉÔØÓÑ Ë ×ÏÚÍÏÖÎÙÍ ÐÒÏ×ÅÒËÁÍ ÉÌÉ ÒÁÚÏÂÒÁÔØÓÑ Ó ÄÏËÕÍÅÎÔÏÏÂÏÒÏÔÏÍ. úÁÈÏÄÉÔÅ ÎÁ ÎÁÛ ÓÁÊÔ - ÕÞÉÔØÓÑ ÓÔÁÌÏ ÇÏÒÁÚÄÏ ÐÒÏÝÅ. é ÍÙ ×ÓÅÇÄÁ ÐÏÍÏÖÅÍ, ÒÁÓÓËÁÖÅÍ, ÏÂßÑÓÎÉÍ É ÐÏÄÂÅÒ£Í ÔÏ, ÞÔÏ ×ÁÍ ÐÏÄÏÊÄ£Ô! [...] Content analysis details: (9.5 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.5 BAYES_99 BODY: Bayes spam probability is 99 to 100% [score: 0.9981] -0.0 SPF_PASS SPF: sender matches SPF record 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [2.146.222.10 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [2.146.222.10 listed in bb.barracudacentral.org] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines

1 0

Problem with dumping offline messages
by Stoyan Mihaylov 06 Sep '12

06 Sep '12

Following examples, I tried to add processing of offline messages. Structure of my system is: SIP client <->Kamailio<->Asterisk<->Kamailio<->SIP client Where Kamailio in above is same server, but SIP clients are different. Now I have no problems with audio, video calls and messages, if both clients are online. All messages received from client are sent to Asterisk, where I can or not process them and then I forward back to Kamailio. When messages came from Asterisk server, then they are sent directly to end user. Sending messages to end user - I call this function, only if message come from allow_source_address(): route[MSILO_MESSAGE]{ if (lookup("location")){ xlog(" Relying"); t_relay(); }else{ xlog("Storing MSILO_MESSAGE $ru-$tu-$ou"); m_store("$ru"); t_relay(); t_reply("202", "Accepted"); }; exit(); } When client register: route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location",0x04)) sl_reply_error(); # MSILO - dumping user's offline messages if (m_dump("$fu")){ xlog("MSILO: offline messages for $fu dumped - if they were \n"); }else{ xlog("MSILO: no offline messages dumped\n"); }; exit; } } I am sending message to offline contact, and I see record in silo table. Record looks perfect. Then offline contact become online, I see in debug MSILO: offline messages for {offlined contact} dumped - if they were But client do not receive nothing. In case all contacts are online, then message go to contact, and is not stored in silo. Setup of msilio is: loadmodule "msilo.so" modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@MY DOMAIN") modparam("msilo","contact_hdr","Contact: registrar@MYDOMAIN:5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","***$rU***$rb***") By the way - I receive in Asterisk offline message. Obviously, I am doing something wrong, but I cant understand what. Any help is welcome. Version of Kamailio is: #~> kamailio -V version: kamailio 3.2.3 (x86_64/linux) 59f87e flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 59f87e compiled on 14:43:47 Jul 19 2012 with gcc 4.5.2

2 1

frequency of dispatcher probing
by Juha Heinanen 05 Sep '12

05 Sep '12

i have defined: modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 3) modparam("dispatcher", "ds_probing_mode", 2) and still i see that dispatcher is pinging inactive node at least once per second: U 2012/09/05 13:47:04.143977 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:04.644185 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:05.143991 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:06.143975 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:06.144056 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... what is that i'm missing in order to make ping interval 10 seconds? -- juha

2 2

Re: [SR-Users] memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

> The 20% initialy loaded is because I load many data with MTREE, > HTABLE,DIALPLAN etc....(5 million rows with MTREE..) > Regarding the PKG, how do i set it? I read a few months ago that the > shared and private memory on version 3.2.2 are the same. > Thanks for the help :-) > > On 09/05/2012 11:44 AM, Uri Shacked wrote: > >* Hi, *>* Thanks, *>* How do i find memory leaks? *>* when my > application starts the it shows the following values: *>* shmem:fragments > = 143898 *>* shmem:freesize = 3446570952 *>* shmem:max_used_size = > 861854768 *>* shmem:real_used_size = 848396344 *>* shmem:total_size = > 4294967296 *>* shmem:used_size = 319676976 *>* I divided the > "real_used_size" with the "total_size" and notice that *>* it is 20%. So, > this is my initial value of the application. *>* After 2 months with > 40,000 calls per day, the relations between these *>* values is 39%. > (doing a reload of the data that occupies 20% failed *>* due to "no > memory") *>* So, 19% occupation of memory is normal for this amount of > calls? or is *>* it a memory leak? if it is a memeory leak, how do i see > where? *Hello, > > Too little information to go on here. A reload might fail mostly because > data from the DB (e.g. with carrierroute case) is first loaded into > private memory (the sql result). Depending of the module there are other > tricks you could do (decrease the number of fetch rows the query returns > etc.). The out of memory condition affects shared or private mem. Try to > use at least 15 MB for PKG mem. > > If the 39% is with traffic, then it's normal, if you don't have any > traffic I still would not worry. The thing that worries me a little is > the value, you are using from the start around 0.8 GB of used shared > memory... this is a lot. From the percentages you provided it looks ok.... > > Marius > >* Uri *>* On 09/05/2012 10:19 AM, Uri Shacked wrote: *>* >/Hi, />/When I > start kamailio it loads some data into memory. />/Now *>* calls are > starting to gather up. How can I check how mach memory *>* />/does my > avp's take? />/BR, />/Uri /Hello Uri, *>* *>* avp's are associated with > the transactions, so they free'd when the *>* transaction stops(and calls > are composed of transactions). This is true *>* for dialog information as > well (dlg_var's etc.). Continuous growth in *>* memory at a constant call > rate (incoming sip requests) might mean a *>* leak. If you need > statistics about the total memory usage you can use *>* `kamctl fifo > get_statistics all` and check for the shmem real_used_size *>* and > max_used_size fields. *>* *>* Cheers, *>* Marius *>* *>* -- *>* Zbihlei > Marius *>* *>* Head of *>* Linux Development Services Romania *>* * > > -- > Zbihlei Marius > > Head of > Linux Development Services Romania > > 1&1 Internet Development srl Tel KA: 754-9152 > Str Mircea Eliade 18 Tel RO: +40-31-223-9152 > Sect 1, Bucuresti mailto: marius.zbihlei at 1and1.ro<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > 71295, Romania > >

1 0

memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, Thanks, How do i find memory leaks? when my application starts the it shows the following values: shmem:fragments = 143898 shmem:freesize = 3446570952 shmem:max_used_size = 861854768 shmem:real_used_size = 848396344 shmem:total_size = 4294967296 shmem:used_size = 319676976 I divided the "real_used_size" with the "total_size" and notice that it is 20%. So, this is my initial value of the application. After 2 months with 40,000 calls per day, the relations between these values is 39%. (doing a reload of the data that occupies 20% failed due to "no memory") So, 19% occupation of memory is normal for this amount of calls? or is it a memory leak? if it is a memeory leak, how do i see where? Uri On 09/05/2012 10:19 AM, Uri Shacked wrote: >* Hi, *>* When I start kamailio it loads some data into memory. *>* Now calls are starting to gather up. How can I check how mach memory *>* does my avp's take? *>* BR, *>* Uri *Hello Uri, avp's are associated with the transactions, so they free'd when the transaction stops(and calls are composed of transactions). This is true for dialog information as well (dlg_var's etc.). Continuous growth in memory at a constant call rate (incoming sip requests) might mean a leak. If you need statistics about the total memory usage you can use `kamctl fifo get_statistics all` and check for the shmem real_used_size and max_used_size fields. Cheers, Marius -- Zbihlei Marius Head of Linux Development Services Romania

2 1

memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, When I start kamailio it loads some data into memory. Now calls are starting to gather up. How can I check how mach memory does my avp's take? BR, Uri

2 1

event_route[dialog:failure]
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, i seems to me that the "event_route[dialog:failure]" does not recognize dialog failures. i use dlg_manage when i recieve the INVITE. i get 301, 302, 402 or send_reply with some +300 responses. on the manage_failure i print the responses, on the event_route[dialog:start] and [dialog:end] i get good results as well. but, the dialog:failure] does not rise.... any ideas? BR, Uri

2 2

LUA Authentication.
by David | StyleFlare 05 Sep '12

05 Sep '12

I think I saw once an example from miconda using LUA for Auth? Does anyone else remember seeing that? I wanted to do a custom Auth using LUA. Thanks in advance for any pointers. David.

4 3

Re: [SR-Users] t_check_status not found in readmes
by Alex Balashov 05 Sep '12

05 Sep '12

It is true. It disappeared from the docs after the 3.0 sip-router convergence. It is present in 'tm' docs in <= 1.5.x, though. The current docs do have t_grep_status(), which seems in many respects identical. I wonder if one is an alias for another now. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Juha Heinanen <jh(a)tutpro.com> wrote: >looks like t_check_status is not documented in any readme file. > >-- juha > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

t_check_status not found in readmes
by Juha Heinanen 05 Sep '12

05 Sep '12

looks like t_check_status is not documented in any readme file. -- juha

1 0

Cassandra module - query question
by Boudewyn Ligthart 03 Sep '12

03 Sep '12

Hi, I would to query the Cassandra database for other information with either CQL or other commands. Could this be done through means of either avpops or sqlops, like an MySQL database. This could be done through an external script but I would like to utilize the connection that is already established. Thanks, Bou

3 4

Why append_branch should be called in a failure route ?
by Vitaliy Aleksandrov 03 Sep '12

03 Sep '12

Below is the example from http://kamailio.org/docs/modules/3.2.x/modules/tm.htm : failure_route[0]{ if (t_check_status("5[0-9][0-9]")){ # I do not like the 5xx responses, # so I give another chance to "foobar.com", # and I drop all the replies to make sure that # they are not forwarded to the caller. t_drop_replies(); rewritehostport("foobar.com"); append_branch(); t_relay(); } } Why do append_branch() is present before t_relay(). I'm asking because I didn't used it in failure routes and call rerouting on various conditions worked great.

1 0

Catch error "status rewrite by UAS" in on_reply
by Alex Hermann 03 Sep '12

03 Sep '12

Hello all, i sometimes get a very late reply from a broken client which triggers the following error 9because the call was already continued in another branch): ERROR: tm [t_reply.c:1193]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Not a problem perse, but i would like to catch this error in the script's on_reply route because I need to prevent some actions on the reply if it will not be forwarded to the UAC. Is there a way to do this? -- Greetings, Alex Hermann

1 0

Celebrating 11 Years
by Daniel-Constantin Mierla 03 Sep '12

03 Sep '12

Hello, eleven years ago, the first piece of code of the project was committed. Last year we organized a public event, for the 10 years celebration, presentations and more details about it at: * http://sip-router.org/10-years-ser/ For the 11th year, I made a news with the summary of the past year - definitively it was lot of work, adding many new features: * http://www.kamailio.org/w/2012/09/kamailio-at-11-years/ Looking forward to the next year celebration, surely it will list another great set of new features! Many thanks to everyone helping the project, from advocating to development! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

2 1

kamailio and clustered database
by Vieri 01 Sep '12

01 Sep '12

Hi, I'm new to Kamailio and would like to set it up with a "circular replication database" on several servers (master-to-master). Before running 'kamdbctl create' I edited kamctlrc and set DBENGINE to MYSQL (I can use either MySQL or PostgreSQL or both - I just need to use a cluster of servers). What would you suggest? Should I set DBENGINE=MYSQL and configure a cluster of databases as described here: http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster-replication-multi-mast… Or does kamailio provide another way to do the same job? What does the db_cluster module do and where/how should I use it? Thanks, Vieri

1 0

weird dp_translate result
by Juha Heinanen 01 Sep '12

01 Sep '12

i have this piece of config code: modparam("dialplan", "attrs_pvar", "$var(plan)") ... xlog("L_INFO", "Translating <$var(calling_number)> based on dpid 4\n"); if (dp_translate("4", "$var(calling_number)")) { xlog("L_INFO", "Found attrs <$var(plan)>\n"); ... dpid 4 has three entries: 109 4 10 1 ^\+1 0 na 107 4 10 1 ^\+358 0 fi 108 4 10 1 ^\+46 0 se thus in each entry attrs field has some two letter value. when the above piece of code is executed, i get to syslog Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Translating <+35892345670> based on dpid 4 Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Found attrs <0> if i restart the proxy, attrs value may change to some other bogus value. any ideas? -- juha

2 10

OpenIMSCore icscf problem
by Siavash Safi 01 Sep '12

01 Sep '12

Hi, I fetched the latest source code from OpenIMSCore SVN repository and generated the debian packages on squeeze. Everything is working fine so far except the icscf module which fails to start. I think that there is something wrong with the configuration. Here is the output messages: root@spawn:~# icscf.sh Listening on udp: xxx.xxx.xxx.xxx [xxx.xxx.xxx.xxx]:5060 tcp: xxx.xxx.xxx.xxx [xxx.xxx.xxx.xxx]:5060 Aliases: tcp: spawn:5060 udp: spawn:5060 *: ims.mydomain.com:* *: icscf.ims.mydomain.com:* ------ Semaphore Arrays -------- key semid owner perms nsems root@spawn:~# 0(4597) ERROR: mk_proxy: could not resolve hostname: "" 0(4597) ERROR: fixup_hostport2proxy: bad host name in URI <> ERROR: error -478 while trying to fix configuration 0(4597) INFO:destroy_diameter_peer(): Bye Bye from C Diameter Peer test As you can see mk_proxy is called with a empty hostname! I have pasted my configuration and more verbose debug output: configuration: http://pastebin.com/nDQM1KLM debug output: http://pastebin.com/d6jh0CHF Any help would be appreciated. -- Regards, Siavash Safi Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

1 0

Kamailio - Nonce validity between Register and Invite
by patrice.bodeven＠orange.com 31 Aug '12

31 Aug '12

Hello, I am working on Kamailio 3.2.2. There is no traffic, only functional test done. Based on the SIP Client used until now (Xlite), the INVITE is systematically authenticated by 407 as there is no Proxy-Authorization in the initial INVITE. Expected/normal behavior. But when using an internal SIP client, the initial INVITE is generated with a Proxy-Authorization containing the nonce value used by the successful registration. In Authorization of Register (the answer is 200OK) Digest username="+33296488922(a)ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:ims.v0.pftest.net", response="4ed51b8e62cf769b19f8c4771e068ad3", cnonce="06440000000529330fe1", qop=auth, nc=00000011 In Proxy-Authorization of Invite Digest username="+33296488922(a)ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:0157361149@ims.v0.pftest.net", response="b247052d6d5b37e0b6677c816390cb77", cnonce="06440000000529330fe1", qop=auth, nc=00000012 As the nonce expire is high (see nonce_expire), I don't understand why Kamailio requests the Authentication of the INVITE by 407. 407 with Proxy-Authenticate: Digest realm="ims.v0.pftest.net", nonce="UDP6pVAzpkWDF+jSlP3zzVYcuNPhY8NQBj62WIA=", qop="auth", stale=true Does it mean that the REGISTER and INVITE are using different Nonce value ? ==> Information on my configuration file. modparam("auth_db", "db_url", DBURLAUTH) modparam("auth", "auth_checks_register", 6) # callid and From TAG modparam("auth", "qop", "auth") # enable qop=auth modparam("auth", "nonce_count", 1) # enable nonce_count support modparam("auth", "nonce_expire", 21600) # Set nonce_expire to 6 hours For register, I do if (!www_authorize("$ru", "subscriber")) { if($? == -2 || $? == -3) { sl_send_reply("403","Forbidden auth ID"); exit; } else { www_challenge("$ru", "17"); exit; } } For invite, I do if (!proxy_authorize("$fd", "subscriber")) { if($? == -2 || $? == -3) { sl_send_reply("403","Forbidden auth ID"); exit; } else { proxy_challenge("$fd", "17"); xlog("L_INFO","INVITE challenged by 407 to :$di:$dp \n"); exit; } } Followed by consume_credentials(); Thanks in advance for your support. Patrice B _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

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invite/ack avps/transactions
by Juha Heinanen 31 Aug '12

31 Aug '12

i tried to access an avp that was set during invite processing while processing ack to 200 ok, but i guess the avp is already gone by that time, because positive ack is its own transaction? what i tried to do is to save somewhere invite's $du and then t_relay ack to the same place without calling loose_route. should i go for htable as storage or is there a better choice? -- juha

2 1

LCR (next_gw) question
by Nord7 31 Aug '12

31 Aug '12

Hello everyone. Configures the module LCR. I can't get him to take the next on the list of priorities gateway if the first does not work. Function "next_gw" capable of doing that? Did anyone realties that? If possible show example. Kamailio 3.3 Regards, Kirill

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spam messages breaking through filters
by Daniel-Constantin Mierla 31 Aug '12

31 Aug '12

Hello, there were some spam messages going through our mailing list filters. I was updating the filters on mailman, but there are new and new ideas coming from this guys. Sorry for the pollution and I wanted to say that these messages are marked as spam by spamassasin, you can filter them locally as well looking for the header X-Spam-Level: *** You can adjust the number of '*'. Some email services do that automatically, like gmail.com for example. Of course, if anyone has better ideas of protection, please write here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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db_cluster together with the registrar module = signal 11
by Øyvind Kolbu 30 Aug '12

30 Aug '12

Hi, I upgraded our system to Kamailio 3.3 to take advantage of the new db_cluster module. Currently we have two registration/location servers each with their own sql-server. Authenticated REGISTER messages are forwarded to the other server, which uses the 0x02 flag to save() in order to store it without generating an extra reply. This works fine. My goal was to use db_cluster to update both databases at once with save() and use the failover mode when reading with lookup(). Have anyone tested a similar setup? Anyhow, here is how I tried to do it: modparam("db_cluster","connection","voip1_data2=>postgres://user:pass@hostname1/voipt1_data2") modparam("db_cluster","connection","voip2_data2=>postgres://user:pass@hostname2/voipt2_data2") modparam("db_cluster", "cluster", "data2=>voip1_data2=8s7p;voip2_data2=9s7p") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "cluster://data2") But I get: : : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 33 : ALERT: <core> [main.c:785]: child process 28275 exited by a signal 11 : ALERT: <core> [main.c:788]: core was generated : INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Same with db_mode = 1 or 3, and either modparam("db_cluster", "cluster", "data2=>voip1_data2=8s9p") or modparam("db_cluster", "cluster", "data2=>voip2_data2=8s9p"). No difference between newly booted phones or phones reregistering. Backtrace from gdb: (gdb) bt #0 0x002fd7da in db_cluster_insert (_h=0xb7d2dc88, _k=0xbf9ac954, _v=0xbf9ac7ec, _n=18) at dbcl_api.c:334 #1 0x00fbb1cc in db_insert_ucontact (_c=0xb5d828d8) at ucontact.c:565 #2 0x00fcfc5d in insert_ucontact (_r=0xfd3a00, _contact=0xb7d3ce90, _ci=0xc5cc80, _c=0xbf9acb84) at urecord.c:478 #3 0x00c55bb0 in add_contacts (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:493 #4 save (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:864 #5 0x00c4dd43 in w_save2 (_m=0xb7d05e40, _d=0xb5d63060 "00ֵ", _cflags=0x0) at reg_mod.c:407 #6 0x080586ad in do_action (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1157 #7 0x0805fe5c in run_actions (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1644 #8 0x080604e1 in run_actions_safe (h=0xbf9adcb8, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1708 #9 0x080f96b7 in rval_get_int (h=0xbf9adcb8, msg=0xbf9ac954, i=0xbf9ad268, rv=0xb7c406cc, cache=0x0) at rvalue.c:920 #10 0x080fcf38 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad268, rve=0xb7c406c8) at rvalue.c:1914 #11 0x080fd009 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad544, rve=0xb7c40a70) at rvalue.c:1922 #12 0x080585fe in do_action (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1121 #13 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1644 #14 0x08058657 in do_action (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1136 #15 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1644 #16 0x0805853d in do_action (h=0xbf9adcb8, a=0xb7bd64d8, msg=0xb7d05e40) at action.c:761 #17 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7bc7128, msg=0xb7d05e40) at action.c:1644 #18 0x08060464 in run_top_route (a=0xb7bc7128, msg=0xb7d05e40, c=0x0) at action.c:1729 #19 0x080e11e4 in receive_msg (buf=0x82c4f20 "REGISTER sip:hometest.voip.uio.no SIP/2.0\r\nVia: SIP/2.0/UDP 193.157.149.56;branch=z9hG4bKb8e88f2eE34DCDDD\r\nFrom: \"kolbutest\" <sip:2549619@hometest.voip.uio.no>;tag=34CE9BA6-70FCC135\r\nTo: <sip:2549619@"..., len=809, rcv_info=0xbf9ade98) at receive.c:209 #20 0x08177f7b in udp_rcv_loop () at udp_server.c:544 #21 0x080b0426 in main_loop () at main.c:1633 #22 0x080b3d02 in main (argc=17, argv=0xbf9ae174) at main.c:2546 Core dumps and debug logs (debug=3) are available. -- Øyvind Kolbu University of Oslo

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Job ad - see details! Sent through Search engine
by serusers＠iptel.org 30 Aug '12

30 Aug '12

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: I would like to take this time to welcome you to our hiring process and give you a brief synopsis of the position's benefits and requirements. If you are taking a career break, are on a maternity leave, recently retired or simply looking for some part-time job, this position is for you. [...] Content analysis details: (6.7 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_PASS SPF: sender matches SPF record 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [189.4.159.143 listed in bb.barracudacentral.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [189.4.159.143 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [189.4.159.143 listed in bl.score.senderscore.com] 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?189.4.159.143>] -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0000] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

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