- sr-users - kamailio.org

t_relay_to_tls cannot be found
by jaflong jaflong 31 Jan '14

31 Jan '14

Can anyone suggest why t_relay_to_tls cannot be found 0(15284) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command t_relay_to_tls i am using kamailio 4.1.1

2 1

Kamailio is not starting when it connects against remote mysql server
by José Seabra 30 Jan '14

30 Jan '14

Hi there, I'm having a strange behavior when i start kamailio, it start processes but after few seconds all processes die, <core> [main.c:1630]: main_loop(): main_loop: Cannot fork Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7354]: DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql:// xxxx:xxxx@10.0.30.15/kamailio Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on '10.0.30.15' (4) Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: uri_db [checks.c:249]: uridb_db_init(): unable to connect to the database Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [sr_module.c:896]: init_mod_child(): init_mod_child(): Error while initializing module uri_db (/usr/local/lib64/kamailio/modules/uri_db.so) Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: ERROR: <core> [pt.c:350]: fork_process(): ERROR: fork_process(): init_child failed for process 5, pid 7335, "udp receiver child=4 sock=10.0.20.7:5060 ( 213.58.213.226:5060)" Jan 30 18:08:54 vx00-lss01 /usr/local/sbin/kamailio[7335]: : <core> [main.c:1630]: main_loop(): main_loop: Cannot fork anyone has idea why it is happening, it was working before and we didn't any changes -- Cumprimentos José Seabra

2 2

Re: [SR-Users] Kamailio behind NAT
by John Smith 30 Jan '14

30 Jan '14

Hello Klaus, I have been making some tests just to be sure the network is not the problem. While using a simpler config file for Kamailio with just the TLS and NAT which does not involve Asterisk in the scenario, calls work properly between softphones. OTOH, using the template provided in the KB referenced in my first post disabling the Asterisk define, calls are connected properly but no audio flows between phones. Now I am in the process of trying to locate where is the problem by comparing how both files handle the NAT support. Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/23/14 08:12 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 23.01.2014 10:29, John Smith wrote: > > Hello Klaus, > > > > I had already two sockets bound each to two independent physical interfaces. I have added the force_send_socket at each rtpproxy > > Just for clarity: > > force_send_socket is for near_end NAT traversal of the SIP signaling, > whereas manage_rtpproxy() is for the NAT traversal (near end and far > end) of the RTP stream. > > > It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? > > If rtpproxy uses only a single listen-IP, then these flags are not > needed. Only if you operate rtpproxy in bridge mode, then you need these > flags. Bridge mode is necessary if you do not have IP routing between > the internal network and the "virtual external" network, or if you want > to bridge between IPv4 and IPv6. > > > Currently audio does not flow back to the softphones, it gets lost at Kamailio. > > Actually it should get lost at rtpproxy. > > Please post a SIP trace: ngrep -Wbyline -q -t -P "" port 5060 > > and post the setup (external + internal IP addresses) (you can send it > privately to me or mangle the IP addresses if they are sensitive) > > regards > Klaus > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Kamailio with mohqueue module
by mark＠brightvoip.co.uk 30 Jan '14

30 Jan '14

Hi, I am developing a system using Kamailio 4.1.1 as a front-end to a group of Asterisk servers. All of the usual functionality (registations, inbound, outbound, internal calls) are fine. We need to add call queueing to the system. I looked at the mohqueue module and read that we can use mohq_process() to send inbound calls to a suitably defined queue, based on the incoming URI. My question is what is the process for managing/automating the call to mohq_retrieve() to send calls to agents? Do I need to track agents with XMPP presence, etc? Thanks, Mark Hall

2 1

tcp_keepalive, tcp_keepidle and tcp_connection_lifetime
by Owais ul Haq 30 Jan '14

30 Jan '14

Hello, I am trying to understand kamailio.cfg parameters. Can some one please help me in understanding what these parameters do exactly when set. tcp_keepalive, tcp_keepidle and tcp_connection_lifetime. I have a general understanding of tcp. Want to know exact role of these parameters in kamailio.cfg. Thanking in advance. Cheers

2 1

htable failed auth banning
by Salvatore Nuzzo 30 Jan '14

30 Jan '14

hey guys. I was trying to add the htable routing rules here: http://preview.tinyurl.com/nomqsls but I need some help figuring our where they should go. See, I really like how this guy bans users for 15 minutes for failing to auth 3 times in a row. Where and how should I add this to my kam config? Any help would be greatly appreciated. Thanks

3 2

Re: [SR-Users] Kamailio behind NAT
by John Smith 30 Jan '14

30 Jan '14

Hello Andrew, I did not know this function. I am using it to determine the direction of the flow. Thank you > ----- Original Message ----- > From: Andrew Pogrebennyk > Sent: 01/24/14 06:51 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 01/23/2014 05:12 PM, Klaus Darilion wrote: > >> It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? > > > > If rtpproxy uses only a single listen-IP, then these flags are not > > needed. Only if you operate rtpproxy in bridge mode, then you need these > > flags. Bridge mode is necessary if you do not have IP routing between > > the internal network and the "virtual external" network, or if you want > > to bridge between IPv4 and IPv6. > > John, > This function can be used to check the direction of every message: > http://kamailio.org/docs/modules/4.0.x/modules/rr.html#idp223296 > You might also need to append the record-route parameters to remember > the flags you have passed to the manage_rtpproxy() initially. > Based on the direction of the request and initial flags you can > determine what flags to use when calling manage_rtpproxy() for a given > in-dialog requests and reply. > > Hope this helps. > Andrew > > _______________________________________________ > 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] Kamailio behind NAT
by John Smith 30 Jan '14

30 Jan '14

Hello Fred, calls are being bridged across two interfaces: one natted from the outside with one public IP and one directly connected to the Asterisk. > ----- Original Message ----- > From: Fred Posner > Sent: 01/24/14 06:55 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 01/24/2014 09:51 AM, Andrew Pogrebennyk wrote: > > On 01/23/2014 05:12 PM, Klaus Darilion wrote: > >>> It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? > >> > >> If rtpproxy uses only a single listen-IP, then these flags are not > >> needed. Only if you operate rtpproxy in bridge mode, then you need these > >> flags. Bridge mode is necessary if you do not have IP routing between > >> the internal network and the "virtual external" network, or if you want > >> to bridge between IPv4 and IPv6. > > > > John, > > This function can be used to check the direction of every message: > > http://kamailio.org/docs/modules/4.0.x/modules/rr.html#idp223296 > > You might also need to append the record-route parameters to remember > > the flags you have passed to the manage_rtpproxy() initially. > > Based on the direction of the request and initial flags you can > > determine what flags to use when calling manage_rtpproxy() for a given > > in-dialog requests and reply. > > > > Hope this helps. > > Andrew > > > > Are the calls being bridged across two interfaces or is the Kamailio > just natted? (or is it both?) > > Fred Posner > The Palner Group, Inc. > 503-914-0999 (direct) > 954-472-2896 (fax) > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Problem Re-writing RURI
by Geoffrey Mina 30 Jan '14

30 Jan '14

I am trying to manually rewrite the RURI for a certain scenario. The RURI is currently: sip:110219546496707@12.12.12.12;npdi;rate=0.000880 I want to re-write so it's just: sip:110219546496707@12.12.12.12 I have the following snippet which isn't working: xlog("L_INFO","sip:$(ruri{uri.user})@$(ruri{uri.host})"); $ru = "sip:$(ruri{uri.user})@$(ruri{uri.host})"; t_relay(); the log spits out exactly what I want, but writing the $ru variable results in: Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "$(ruri{uri.host})" Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "$(ruri{uri.host})" Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [t_fwd.c:1773]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [tm.c:1502]: _w_t_relay_to(): ERROR: w_t_relay_to: t_relay_to failed Jan 29 18:59:36 dev-sipgateway /usr/local/sbin/kamailio[23042]: ERROR: tm [t_reply.c:1067]: run_failure_handlers(): ERROR: run_failure_handlers: Error in run_top_route Any hint would be greatly appreciated.

2 1

Notice: changes to apt repositories
by Daniel-Constantin Mierla 29 Jan '14

29 Jan '14

Hello, the current server hosting the apt repositories for debian and ubuntu is scheduled for replacement. It hasn't been updated with new packages in a while. Courtesy of Sipwise, a new build system is under development and should be available in the near future. Meanwhile, we plan to clone the existing packages to another server and have them available till the new system is up and running. As all these involve operations requiring DNS change and file transfer, the repositories might be unavailable from time to time. Many thanks to Jon Bonilla that took care of building the deb packages so far and Inaki Baz Castillo for hosting the afferent repositories. Note also that all Kamailio packages for latest version are also available via the official repositories from Debian unstable (sid) - thanks to Victor Seva for the work on it. Perhaps testing will get them soon and with the next stable version, should be no longer needed to use our repositories by users of Debian. We plan to keep ours as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Contents of sr-users digest...
by Stuart Feeser 28 Jan '14

28 Jan '14

Regarding the sem_post problem, here is an old email from Tomasz Gebarowski which is interesting because it states that he had a sem_post issue with cdp.so which he solved by modifying the modules/cdp/Makefile. Now if we skip forward to right now, I notice that cdp.so COMPILE and LINKS for me with no errors and I also notice that Tomasz' fix is in place, but other ims modules that also use semaphore do NOT link properly for me in ubuntu and are exhibiting the same *undefined symbol: sem_post* error Tomasz had years ago in his ubuntu compiles. I think (hope) this may be more than coincidence that these things are related??? I have tried Tomasz' fix with an ubuntu compile but I still get the sem_post error. I still think he is on to something here and I am NOT modifying the Makefile correctly. Any suggestions on specifics to test this theory are greatly welcome. Here is Tomansz' email..... ---------------------------------------------------------------------------------------------------------------------- https://lists.berlios.de/pipermail/openimscore-users/2011-December/007219.h… [OpenIMSCore-Users] PROBLEM WITH ICSCF AND SCSCF STARTING Tomasz Gebarowski gebarowski at gmail.com Thu Dec 29 22:19:55 CET 2011 Previous message: [OpenIMSCore-Users] Fwd: How are you... Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hello, I've just googled some old post from October 2011, describing an issue with crashing I-CSCF and S-CSCF. The log message was something like: 0(2142) ERROR: load_module: could not open module /opt/OpenIMSCore/ser_ims/ modules/cdp/cdp.so>: /opt/OpenIMSCore/ser_ims/modules/cdp/cdp.so: undefined symbol: sem_post I managed to fix this issue on Ubuntu 11.10. It seems that UBUNTU, required linking cdp.so against pthread. Below you can find a simple patch: Index: modules/cdp/Makefile =================================================================== --- Makefile (revision 1175) +++ Makefile (working copy) @@ -20,7 +20,7 @@ endif DEFS += -I/usr/include/libxml2 -LIBS += -L$(LOCALBASE)/lib -lxml2 -lrt +LIBS += -L$(LOCALBASE)/lib -lxml2 -lrt -lpthread include ../../Makefile.modules =================================================================== Regards, Stuart Feeser

1 0

Kamailio with Asterisk
by Chirag Desai 28 Jan '14

28 Jan '14

Hi all, I'm wondering if someone can help me get a little unstuck. I followed the guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And I now have kamailio running, but I have two problems. Number 1 When I do kamctl add user password and make two users, they register fine. But any attempt to call between the users times out. Number 2 Anytime I have my kamailo.cfg file begin as #!KAMAILIO #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_ASTERISK I am unable to start kamctl. I get the following error. ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed If I run kamailio I get ERROR: bad config file (3 errors) The moment I remove the define lines at the start, kamctl starts fine, but I can't authenticate with asterisk. Any ideas? Thanks Chirag

3 2

ser-0.9.6 bug report
by David Binderman 28 Jan '14

28 Jan '14

Hello there, I just ran the static analyser "cppcheck" over the source code of ser-0.9.6 It said many things, including 1. [parser/parse_uri.c:566] -> [parser/parse_uri.c:568]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? Source code is case 'r': case 'R': b=p; state=PR2_R; default: state=URI_PARAM_P; Suggest add missing break. 2. > [parser/parse_uri.c:624] -> [parser/parse_uri.c:624]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? > [parser/parse_uri.c:628] -> [parser/parse_uri.c:628]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? > [parser/parse_uri.c:631] -> [parser/parse_uri.c:631]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? > [parser/parse_uri.c:636] -> [parser/parse_uri.c:636]: (warning) Variable 'state' is reassigned a value before the old one has been used. 'break;' missing? Duplicates. Regards David Binderman

2 1

Replacing DR Module with Perl Script
by Nick Cameo 27 Jan '14

27 Jan '14

Hello Everyone, We are currently using the DR module for routing and failover, and would like to replace it with equivalent functionality using perl script. What we would like is to: * Order the gateways in a strict list given some constraints (Completed) * Build the RURI with the gateway IP, and prefix from the dr_gateways table (Similar to what do_routing does with the FROM tag and dr_gateways information) * Failover to the next gateway in strict order. I would really like to know what our options are, and the most graceful/scalable way of implementing this. PS An example would be highly appreciated! Kind Regards, Nick

2 2

undefined symbol: sem_post
by Stuart Feeser 27 Jan '14

27 Jan '14

My exact problem is described here,<http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html> but the thread never explains how to solve the problem, *"here" = * http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html *Description of the issue: * 1. Installing Kamailio 4.1 (latest download) on a *64 bit* Linux Mint version 16 Petra 2. The system compiles cleanly. I have included MAKE and Make Install info below 3. *ims_registrar_pcscf.so will not load* 4. * ims_qos.so will not load* 5. As an FYI, cdp.so successfully loads. I mention this because cdp.so uses semaphore but does not have the sem_post problem. I compared cdp.so with ims_registrar.pcscf.soMakefiles to see if there was something obvious, but no luck. 6. As an FYI, I get the exact same error from a debian apt-get install, so using the canned Debian solution won't work for me either. 7. The config file I used is the one included with the system which I completed with some info just for testing. I have attached it purely as FYI but I don't think that is the issue. *Parts of the syslog messages relevant for issue* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_qos.so>: /lib64/kamailio/modules/*ims_qos.so:* *undefined symbol: sem_post* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_registrar_pcscf.so>: /lib64/kamailio/modules/ *ims_registrar_pcscf.so*: *undefined symbol: sem_post* *Relevent section of MAKE* CC (gcc) [M ims_qos.so] cdpeventprocessor.o CC (gcc) [M ims_qos.so] mod.o CC (gcc) [M ims_qos.so] rx_aar.o CC (gcc) [M ims_qos.so] rx_asr.o CC (gcc) [M ims_qos.so] rx_authdata.o CC (gcc) [M ims_qos.so] rx_avp.o CC (gcc) [M ims_qos.so] rx_str.o CC (gcc) [M ims_qos.so] stats.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so CC (gcc) [M ims_registrar_pcscf.so] async_reginfo.o CC (gcc) [M ims_registrar_pcscf.so] reg_mod.o CC (gcc) [M ims_registrar_pcscf.so] save.o CC (gcc) [M ims_registrar_pcscf.so] service_routes.o CC (gcc) [M ims_registrar_pcscf.so] ul_callback.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so CC (gcc) [M ims_registrar_scscf.so] api.o CC (gcc) [M ims_registrar_scscf.so] common.o CC (gcc) [M ims_registrar_scscf.so] config.o CC (gcc) [M ims_registrar_scscf.so] cxdx_avp.o CC (gcc) [M ims_registrar_scscf.so] cxdx_callbacks.o CC (gcc) [M ims_registrar_scscf.so] cxdx_sar.o CC (gcc) [M ims_registrar_scscf.so] lookup.o CC (gcc) [M ims_registrar_scscf.so] path.o CC (gcc) [M ims_registrar_scscf.so] registrar_notify.o CC (gcc) [M ims_registrar_scscf.so] reg_mod.o CC (gcc) [M ims_registrar_scscf.so] regpv.o CC (gcc) [M ims_registrar_scscf.so] regtime.o CC (gcc) [M ims_registrar_scscf.so] reply.o CC (gcc) [M ims_registrar_scscf.so] rerrno.o CC (gcc) [M ims_registrar_scscf.so] save.o CC (gcc) [M ims_registrar_scscf.so] server_assignment.o CC (gcc) [M ims_registrar_scscf.so] sip_msg.o CC (gcc) [M ims_registrar_scscf.so] stats.o CC (gcc) [M ims_registrar_scscf.so] userdata_parser.o CC (gcc) [M ims_registrar_scscf.so] usrloc_cb.o ---------------------------------------------------------------------------------------------------- *Make Install* LD (gcc) [M ims_icscf.so] ims_icscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_icscf.so install -m 755 ims_icscf.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_qos.so install -m 755 ims_qos.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_registrar_pcscf.so install -m 755 ims_registrar_pcscf.so //lib64/kamailio/modules

1 0

undefined symbol: sem_post
by Stuart Feeser 27 Jan '14

27 Jan '14

My exact problem is described here,<http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html>but the thread never explains how to solve the problem, *"here" = * http://lists.kamailio.org/pipermail/sr-users/2013-January/076554.html *Description of the issue: * 1. Installing Kamailio 4.1 (latest download) on a *64 bit* Linux Mint version 16 Petra 2. The system compiles cleanly. I have included MAKE and Make Install info below 3. *ims_registrar_pcscf.so will not load* 4. * ims_qos.so will not load* 5. As an FYI, cdp.so successfully loads. I mention this because cdp.so uses semaphore but does not have the sem_post problem. I compared cdp.so with ims_registrar.pcscf.so Makefiles to see if there was something obvious, but no luck. 6. As an FYI, I get the exact same error from a debian apt-get install, so using the canned Debian solution won't work for me either. 7. The config file I used is the one included with the system which I completed with some info just for testing. I have attached it purely as FYI but I don't think that is the issue. *Parts of the syslog messages relevant for issue* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_qos.so>: /lib64/kamailio/modules/*ims_qos.so:* *undefined symbol: sem_post* Jan 26 22:03:21 PC03 kamailio: ERROR: <core> [sr_module.c:599]: load_module(): ERROR: load_module: could not open module </lib64/kamailio/modules/ims_registrar_pcscf.so>: /lib64/kamailio/modules/ *ims_registrar_pcscf.so*: *undefined symbol: sem_post* *Relevent section of MAKE* CC (gcc) [M ims_qos.so] cdpeventprocessor.o CC (gcc) [M ims_qos.so] mod.o CC (gcc) [M ims_qos.so] rx_aar.o CC (gcc) [M ims_qos.so] rx_asr.o CC (gcc) [M ims_qos.so] rx_authdata.o CC (gcc) [M ims_qos.so] rx_avp.o CC (gcc) [M ims_qos.so] rx_str.o CC (gcc) [M ims_qos.so] stats.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so CC (gcc) [M ims_registrar_pcscf.so] async_reginfo.o CC (gcc) [M ims_registrar_pcscf.so] reg_mod.o CC (gcc) [M ims_registrar_pcscf.so] save.o CC (gcc) [M ims_registrar_pcscf.so] service_routes.o CC (gcc) [M ims_registrar_pcscf.so] ul_callback.o make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so CC (gcc) [M ims_registrar_scscf.so] api.o CC (gcc) [M ims_registrar_scscf.so] common.o CC (gcc) [M ims_registrar_scscf.so] config.o CC (gcc) [M ims_registrar_scscf.so] cxdx_avp.o CC (gcc) [M ims_registrar_scscf.so] cxdx_callbacks.o CC (gcc) [M ims_registrar_scscf.so] cxdx_sar.o CC (gcc) [M ims_registrar_scscf.so] lookup.o CC (gcc) [M ims_registrar_scscf.so] path.o CC (gcc) [M ims_registrar_scscf.so] registrar_notify.o CC (gcc) [M ims_registrar_scscf.so] reg_mod.o CC (gcc) [M ims_registrar_scscf.so] regpv.o CC (gcc) [M ims_registrar_scscf.so] regtime.o CC (gcc) [M ims_registrar_scscf.so] reply.o CC (gcc) [M ims_registrar_scscf.so] rerrno.o CC (gcc) [M ims_registrar_scscf.so] save.o CC (gcc) [M ims_registrar_scscf.so] server_assignment.o CC (gcc) [M ims_registrar_scscf.so] sip_msg.o CC (gcc) [M ims_registrar_scscf.so] stats.o CC (gcc) [M ims_registrar_scscf.so] userdata_parser.o CC (gcc) [M ims_registrar_scscf.so] usrloc_cb.o ---------------------------------------------------------------------------------------------------- *Make Install* LD (gcc) [M ims_icscf.so] ims_icscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_icscf.so install -m 755 ims_icscf.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_qos.so] ims_qos.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_qos.so install -m 755 ims_qos.so //lib64/kamailio/modules make[2]: `libkcore.so.1.0' is up to date. make[2]: `libkamailio_ims.so.0.1' is up to date. LD (gcc) [M ims_registrar_pcscf.so] ims_registrar_pcscf.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch //lib64/kamailio/modules/ims_registrar_pcscf.so install -m 755 ims_registrar_pcscf.so //lib64/kamailio/modules

1 0

alias_db DID regex?
by Pete Ashdown 26 Jan '14

26 Jan '14

Is there a way to use regex or Asterisk style numerical patterns in alias_db or any other db routing modules? Compressing a block of a hundred DID numbers to one entry would be really nice.

4 6

direct2rupayan@gmail.com has indicated you're a friend. Accept?
by direct2rupayan＠gmail.com 25 Jan '14

25 Jan '14

1 0

Static forward, reverse proxy
by Diego Woitasen 25 Jan '14

25 Jan '14

Hi, I need to configure Kamailio to act as reverse/frontend proxy, with static routing using rtpproxy. It will receive the SIP messages using UDP and must forward the traffic to the backends using TCP. The only IP address that the endpoints must see is the IP address of Kamailio. I've been looking at different examples on the web but I can't found something similar and I can't figure out how to start reading the official docs. If anyone has an example, doc or a hint it will be welcome. Regards, Diego -- Diego Woitasen Linux and Open Source solutions architect at www.vhgroup.net

1 0

Re: [SR-Users] Kamailio behind NAT
by John Smith 24 Jan '14

24 Jan '14

Hello Klaus, I had already two sockets bound each to two independent physical interfaces. I have added the force_send_socket at each rtpproxy It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? Currently audio does not flow back to the softphones, it gets lost at Kamailio. Thank you for your help > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/23/14 12:26 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > Am 21.01.2014 17:33, schrieb John Smith: > > The next test has been to comment out the rtpproxy_manage at NATMANAGE function and to put it both at route[RELAY] and onreply(route) following your post in this list from January 2013:http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html. > > > > Now the media flows from Phone1 to Kamailio, from Kamailio to Asterisk and back, but it gets stuck at Kamailio. I cannot see it flow towards the public IP of the Phone2. > > > > The force_send_socket you used could be of any use here? > That's what I recommend: > > - use 2 sockets, one for communication with internal nodes, one for > external clients > - in your Kamailio config check the direction of every message: i->e or > e->i (for requests and responses). Depending on the direction set the > proper IP when calling manage_rtpproxy and force the send socket: > > regards > Klaus > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

4 3

pv_printf buffer overflow
by Aman Agarwal 24 Jan '14

24 Jan '14

I am running Kamailio-4.1.0 and I am getting this error If I try to parse JSON string longer than 1024 characters. ERROR: <core> [pvapi.c:1338]: pv_printf(): no more space for spec value ERROR: <core> [pvapi.c:1347]: pv_printf(): buffer overflow -- increase the buffer size... ERROR: <core> [sr_module.c:1696]: get_str_fparam(): Could not convert the PV-formated string to str ERROR: json [json_funcs.c:42]: json_get_field(): cannot get json string value I searched online for a fix and it was suggested to increase the PV_PRINT_BUF_SIZE in pvar.c. The source code for Kamailio 4.1.0 does not contain this file. I did not find this constant in other files too . Can some please suggest how to increase the buffer size? Thanks

2 1

Replicate registration
by Volkan Oransoy 23 Jan '14

23 Jan '14

Hi all, I am trying to setup an active-active pair of sip proxies. I have connected two kamailio boxes to the same PostgreSQL database and my usrloc db_mode is 3. I can see the registration data in the database but "ul show" outputs of two device is not same. I think writing registration data to the same db is not enough. I have found a method to replicate the registraiton process. With t_replicate, I synchronize REGISTER requests to both boxes. But I had to disable shared database because of uniqe id validations. route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > } > if (!save("location")) > sl_reply_error(); > t_replicate("sip:10.1.1.2:5060;transport=udp"); > exit; > } > } My media servers randomly sends calls to SIP proxies. If Media server send the call to the original registrar then the replicated, call initiates normally. But when UA registered in first proxy, and Media server sends the call to the send one, even if second one has UA in location table, call does not initiate. Does anyone have an idea about this kind of an active active setup? Thanks in advance. Best regards, /Volkan

4 4

P_usrloc - expired locations
by Helena Garcia-Nieto 23 Jan '14

23 Jan '14

Hi All, I am testing the p_usrloc module in order to write the location information of my kamailo server (Kamailio is on version 4.0.2) in two db As first test both DB are on the same machine, as well as the master db. On final version master and one db would probably be on the same machine and the second db on a different one. I have read on the p_usrloc module the following note: The p_usrloc module still has some missing feature, like automatic expiry of contacts and dumping of all users via the fifo cmd. And I can sadly see that the locations are not being deleted automatically after expiring. My users are mobile apps, that are programmed to be always logged in, and reconnecting to sip and IM servers if the devices detects a drop on the connection (bad wi-fi connection), or the user just move between different 3G cells. So the locations are most likely not going to be properly "unregistered" (expire 0) Have anyone found a solution or can give me some advice on how can I force the expired locations to be erased from the db, just as the usrloc module does? Thanks in advanced! Helena

2 5

Kamailio with extern Presence and XCAP server
by mohammed alyaseen 23 Jan '14

23 Jan '14

Hi all, Since 2 weeks I am trying to get Kamailio presence and xcap servers working. I am testing with RCS clients (native android clients). Somehow the presence server of kamailio doesn't like some subscribe events of RCS for example "Event: xcap-diff;path="resource-lists/users/sip:22222@kamailio.test/index", when RCS sends subscribe with this event, kamailio returns "Bad Event"... Another problem is some AUIDs used by RCS in the HTTP GETs for example ext-resource-lists. Kamailio doesn't support such as AUIDs and replies with not supported auid error.... Now I want to use extern presence and xdms servers, which run on the same virtual machine, where Kamailio is running. What should I do to achieve this? Any advice would be appreciated... Thanks Medo

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 23 Jan '14

23 Jan '14

Hi Shankar, I will setup and test case and try to reproduce as soon as I can. I suspect I will be able to get back to you early next week only Cheers Jason On Thu, Jan 23, 2014 at 8:48 AM, Shankar <shankar.rk(a)plintron.com> wrote: > Hi, > > > > From our repeated load tests what we can conclude irrespective of the > number of simultaneous calls, there is always this error “t_continue: > transaction not found” occurring. > > > > If I run say 20 cps, then after running 5000 calls, we observe exactly 20 > calls failing with the above error. We doubt that there is something > happening during a particular point in time (for a second) which impacts > the saving of those new transactions into shared memory. > > > > For 10 cps run, we observe exactly 10 call failures. We repeated with > different cps and found that error is exactly equal to the cps being run. > > > > Any configuration we are missing. Anyone can help? > > > > Regards, > > Shankar > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Tuesday, January 21, 2014 3:09 PM > *To:* 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express > Router (SER) - Users Mailing List' > *Subject:* FW: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Jason, > > > > Below is our config, > > > > route[LOCATION] { > > if(is_method("INVITE")) > > { > > if(!route(FROMCSCF)) > > { > > setflag(FLT_ACC); > > setflag(FLT_ACCFAILED); > > dlg_manage(); > > dlg_setflag("4"); > > > > async_route("RESUME", "1"); > > > > exit; > > } > > } > > } > > > > route[RESUME] > > { > > route(TO_LOCATION); // here t_relay to REGISTRAR is > done for user lookup. > > exit; > > } > > > > Regards, > > Shankar > > > > > > Date: Tue, 21 Jan 2014 11:14:21 +0200 > > From: Jason Penton <jason.penton(a)smilecoms.com> > > To: "Kamailio (SER) - Users Mailing List" > > <sr-users(a)lists.sip-router.org> > > Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > Message-ID: > > <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > We use it heavily, but not using the async module - we use it directly > from the IMS code. > > > > Can you please provide your config (or a relevant snippet) file so I can > see what exactly you are testing/trying to do > > > > Cheers > > jason > > > > *From:* Shankar [mailto:shankar.rk@plintron.com <shankar.rk(a)plintron.com>] > > *Sent:* Tuesday, January 21, 2014 2:25 PM > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi, > > > > Anyone who had used t_suspend() and t_continue() can share the performance > details? > > > > I tried async module with one sec sleep time. I tried only 5 calls per > second but still it was not successful. After sometime I see below logs, > > > > Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: > t_continue(): ERROR: t_continue: transaction not found > > Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times > > Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times > > Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times > > > > Any configuration changes can help here? > > > > Regards, > > Shankar > > > > *From:* Shankar [mailto:shankar.rk@plintron.com <shankar.rk(a)plintron.com>] > > *Sent:* Wednesday, January 15, 2014 1:26 PM > *To:* 'Jason Penton' > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Jason, > > > > I am using 4.0.2 > > > > Regards, > > Shankar > > > > *From:* Jason Penton [mailto:jason.penton@smilecoms.com<jason.penton(a)smilecoms.com>] > > *Sent:* Wednesday, January 15, 2014 1:21 PM > *To:* Shankar > *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Subject:* Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Shankar, > > > > What version of Kamailio are you running? Kamailio -V > > > > Cheers > > Jason > > > > On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: > > Hi Jason, > > > > Please find below my response inline, > > > > > > I have some questions for you as we have used suspend/continue quite a lot > in the IMS code and don't have any leaks. > > > > Firstly, why are you using pkg_mem for your hash_id and label? Remember > that you will be in 2 different processes in the suspend and continue > portions of the code... so pkg_mem will not work - you should use shm_mem > instead. > > > > [Shankar] We use pkg_mem because we are invoking t_continue from the same > process ( using thread ). > > > > Secondly, how are you using top to tell that you have a leak? Kamailio's > memory is internally managed. > > > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the shared > usage of shared memory for the process. > > > > Cheers > > Jason > > > > > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > > > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com<shankar.rk(a)plintron.com> > ] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > > > > > > -- > > > > *Jason Penton* > > *Senior Manager: Applications and Services* > > *Smile Communications Pty (Ltd)* > > *Mobile:* > > +27 (0) 83 283 7000 > > *Skype:* > > jason.barry.penton > > jason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> > > www.smilecoms.com > > > > This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer > > > > > > > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

Re: [SR-Users] kamailio-4.0.2 core issue
by Daniel-Constantin Mierla 23 Jan '14

23 Jan '14

Hello, can you give the output in gdb for: bt full Also, do you have 'listen' parameters or kamailio is listening on all interfaces? It is recommended to upgrade to latest version 4.0.x, to be sure there is not a side effect of a fixed issues. Cheers, Daniel On 17/01/14 13:50, Surendra wrote: > Hi, > > We are facing a core issue in 4.0.2 version. Please find the GDB > trace below > > #0 udp_rcv_loop () at udp_server.c:471 > 471 fromlen=sockaddru_len(bind_address->su); > (gdb) bt > #0 udp_rcv_loop () at udp_server.c:471 > #1 0x0000000000475c14 in main_loop () at main.c:1851 > #2 0x000000000047a98b in main (argc=<value optimized out>, > argv=<value optimized out>) at main.c:2939 > > > We have simulated this so many times in load mode. > Machine details: Linux PLT-RA-RD-W167A.localdomain 2.6.18-308.el5 #1 > SMP Fri Jan 27 17:17:51 EST 2012 x86_64 x86_64 x86_64 GNU/Linux > > Can anyone help me on this > > > Regards > Surendra > > > > > _______________________________________________ > 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

1 0

Re: [SR-Users] SIP Update multi-homed
by Keith 23 Jan '14

23 Jan '14

Hi, Thanks for the replies so far, good news is I got it working by switching off mhomed (mhomed=0). However this seems to have broken my dispatcher as the SIP OPTIONS probing is being sent from the wrong interface! Any ideas how to set where dispatcher sends these from?? Cheers Keith

2 1

Re: [SR-Users] Kamailio behind NAT
by John Smith 23 Jan '14

23 Jan '14

Hello Klaus, I have changed the rtpproxy_manage() in the route(NATMANAGE) to no avail. The c= in the Invite that arrives to Asterisk still points to the public IP of the Phone1. Code is as follows: route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (src_ip == ip.pbx.priva.te) { rtpproxy_manage(); } else { rtpproxy_manage("","ip.kamailio.priva.te"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } The next test has been to comment out the rtpproxy_manage at NATMANAGE function and to put it both at route[RELAY] and onreply(route) following your post in this list from January 2013: http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html. Now the media flows from Phone1 to Kamailio, from Kamailio to Asterisk and back, but it gets stuck at Kamailio. I cannot see it flow towards the public IP of the Phone2. The force_send_socket you used could be of any use here? Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/21/14 07:25 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > Yes, when calling rtpproxy (whatever function you use, > manage/offer/answer/force), every function accepts as second parameter > the IP address which should be written into the SDP c=... line. > > So, detect the direction (eg based on source-ip) and put the respective > internal or external IP address into the SDP. > > Of course the NAT must be a static 1:1 NAT without any PAT. > > Btw: you have the same problem with some SIP headers, like Record-Route > or Via. Kamailio has to insert the respective public/private IP, > depending on where to send the message. If Kamailio uses only a single > socket, then you have to do this manually. The automatic approach is to > configure 2 listen=... sockets. One for the internal side, and one for > the external side. On the "external" socket you add the "advertise" > option - then Kamailio will automatically put the correct IP addresses > in RR and Via, eg: > > # public IP address of Kamailio/rtpproxy: 3.3.3.3 > # internal socket > listen=udp:10.10.0.2 > # external socket > listen=udp:10.10.0.3 advertise 3.3.3.3:5060 > mhomed=0 > > You could use separate subnets for the 2 sockets, or you could also use > only a single IP address with 2 different ports for the 2 sockets. > > > regards > Klaus > > > On 21.01.2014 14:50, John Smith wrote: > > The intended setup is as you described in option a. > > > > And the behaviour of rewriting SDP using private/public ports is what I understood the rtpproxy_manage was tasked to do. > > > > As it is not working the way I expected, what is the best way to proceed? To check if the IP is from the outside and then rewrite via rtpproxy_offer in the NATMANAGE block? > > > > Thank you > > > >> ----- Original Message ----- > >> From: Klaus Darilion > >> Sent: 01/21/14 05:25 AM > >> To: Kamailio (SER) - Users Mailing List > >> Subject: Re: [SR-Users] Kamailio behind NAT > >> > >> On 21.01.2014 13:24, John Smith wrote: > >>> I might be making wrong assumptions regarding this traffic flow. Is that correct? > >> > >> That depends on your policy. It is up to you to define how RTP should be > >> routed. There are basically 2 choices: > >> > >> a) RTP from clients is handled by rtpproxy: > >> > >> phone1 <-nat-> rtpproxy <--> Asterisk <--> rtpproxy <-nat-> phone2 > >> > >> In this case, only the private IP of Kamailio and rtpproxy (can be the > >> same IP address) must be mapped to a public IP address. > >> > >> > >> b) RTP directly to Asterisk: > >> > >> phone1 <-nat-> Asterisk <-nat-> phone2 > >> > >> In this case, the private IPs of Kamailio and Asterisk must be mapped to > >> a public IP address. > >> > >> > >> When using version a) you have to make sure to set the proper IP address > >> in the SDP. For example, SDPs in messages from Kamailio to the phone > >> must contains the PUBLIC IP of rtpproxy in the c=... line. SDPs in > >> messages from Kamailio to Asterisk must contain the PRIVATE IP of > >> rtpproxy in the c=... line. > >> > >> regards > >> Klaus > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > 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] FW: Regd. t_suspend() and t_continue()
by Shankar 23 Jan '14

23 Jan '14

Hi, >From our repeated load tests what we can conclude irrespective of the number of simultaneous calls, there is always this error "t_continue: transaction not found" occurring. If I run say 20 cps, then after running 5000 calls, we observe exactly 20 calls failing with the above error. We doubt that there is something happening during a particular point in time (for a second) which impacts the saving of those new transactions into shared memory. For 10 cps run, we observe exactly 10 call failures. We repeated with different cps and found that error is exactly equal to the cps being run. Any configuration we are missing. Anyone can help? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

pdd value
by Oliver Roth 22 Jan '14

22 Jan '14

Hi all Question about an additional cdr information. We need to get the pdd value - means a time value till the user gets a ringback or an error- to track the delay, till a connection is done --> quality issue for customer. 5.6.1 Definition of Post Dial Delay Post Dial Delay (PDD) is experienced by the originating customer as the time from the sending of the final dialled digit to the point at which they hear ring tone or other in-band information. Where the originating network is required to play an announcement before completing the call then this definition of PDD excludes the duration of such announcements. How can this be done in Kamailio? We use the carrierroute module with different fallback carriers. Kamailio version 3.3.5 Thanks in advance Oli

2 1

Re: [SR-Users] SIP Update multi-homed
by Keith 21 Jan '14

21 Jan '14

Hi All, Thanks for all the advice, I managed to sort this one (I hope!). For anyone else interested I had to add some code at the start of the routing to look for a SIP UPDATE request. If it was a request then to set the $fs (forced socket) value to equal $Ri (Received interface) value. Once this was set it now holds the call open for longer than 30 seconds, fingers crossed it's all good! Thanks again, hope this is of some help to someone else. Keith

1 0

Re: [SR-Users] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

The intended setup is as you described in option a. And the behaviour of rewriting SDP using private/public ports is what I understood the rtpproxy_manage was tasked to do. As it is not working the way I expected, what is the best way to proceed? To check if the IP is from the outside and then rewrite via rtpproxy_offer in the NATMANAGE block? Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/21/14 05:25 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 21.01.2014 13:24, John Smith wrote: > > I might be making wrong assumptions regarding this traffic flow. Is that correct? > > That depends on your policy. It is up to you to define how RTP should be > routed. There are basically 2 choices: > > a) RTP from clients is handled by rtpproxy: > > phone1 <-nat-> rtpproxy <--> Asterisk <--> rtpproxy <-nat-> phone2 > > In this case, only the private IP of Kamailio and rtpproxy (can be the > same IP address) must be mapped to a public IP address. > > > b) RTP directly to Asterisk: > > phone1 <-nat-> Asterisk <-nat-> phone2 > > In this case, the private IPs of Kamailio and Asterisk must be mapped to > a public IP address. > > > When using version a) you have to make sure to set the proper IP address > in the SDP. For example, SDPs in messages from Kamailio to the phone > must contains the PUBLIC IP of rtpproxy in the c=... line. SDPs in > messages from Kamailio to Asterisk must contain the PRIVATE IP of > rtpproxy in the c=... line. > > regards > Klaus > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Using advertised IP address in manage_rttproxy would work with unpatched rtpproxy? Thank you > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/21/14 05:18 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > On 21.01.2014 12:27, Fred Posner wrote: > > With a patched version of rtpproxy you can advertise your private ip. > > > > http://www.fredposner.com/voip/1457/kamailio-behind-nat/ > > Aha, nice. Haven't known of this one. > > I always specified the "adverstised IP address" when calling > manage_rtpproxy(). That should work too. > > regards > Klaus > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Hello Kostas, the idea is to hide the Asterisk completely behind Kamailio, so I work only with the public IP of Kamailio and its SIP/RTP ports. There are no Asterisk RTP ports enabled, while the Kamailio RTP are already open. Thank you > ----- Original Message ----- > From: meres > Sent: 01/21/14 05:06 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > Hi John, > > rtpproxy is not enough if you are using asterisk in your environment. > You have to check that asterisk is configured to work with NAT, otherwise you will experience audio problems. > Are the asterisk RTP ports enabled/forwarded on your firewall? > > Regards, > > Kostas > > On Jan 21, 2014, at 2:24 PM, John Smith <jsmith.15(a)mail.com> wrote: > > > Hi Fred, > > > > I have followed your HOWTO and the scenario remains exactly the same. > > > > I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP to Asterisk IP, and back directly to Phone2 public IP. > > > > I might be making wrong assumptions regarding this traffic flow. Is that correct? > > > > Thank you > > > > _______________________________________________ > > 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] Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Hi Fred, I have followed your HOWTO and the scenario remains exactly the same. I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP to Asterisk IP, and back directly to Phone2 public IP. I might be making wrong assumptions regarding this traffic flow. Is that correct? Thank you

3 3

Kamailio behind NAT
by John Smith 21 Jan '14

21 Jan '14

Hello, I am currently deploying one Kamailio behind NAT with one Asterisk as explained in the Asipto KB (Kamailio 4.0.x and Asterisk 11.3.0 using Asterisk Database http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ). The structure is deployed as described in that document, with the only addition of one NAT between Kamailio and Internet: Phone ———————> Nat ———————> Kamailio ——————> Asterisk I have declared the private IP with the advertise option in order to support the NAT, enabled WITH_NAT and I have installed rtpproxy using standard Debian package configured as rtpproxy -l public_ip_ -s udp:localhost:7722 After setting up two phones which register correctly at Asterisk, I have no audio at all. By placing tcpdumps between nodes I see at Kamailio node both audio from public IP to internal Kamailio IP and from the latter to the Asterisk IP. In Asterisk I see audio coming from the Kamailio private IP and then back to the public IP of the phone. My guess is that audio should flow back into Kamailio and then to the phone, not directly from Asterisk as it is right now. Can anyone hint at where I am wrong? Thank you

3 2

Re: [SR-Users] Regd. t_suspend() and t_continue()
by Shankar 21 Jan '14

21 Jan '14

Hi Daniel, Please find attached the log as requested. I ran 10 cps to reproduce it faster. Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

SIP Update multi-homed
by Keith 21 Jan '14

21 Jan '14

Hi, I have a kamailio server which is responding to SIP updates but sending from the wrong IP. I have multiple IPs in the same subnet on the same NIC. Is there anyway to say send update back out of the interface it was received on? Thanks. Keith

3 2

Re: [SR-Users] SIP Update multi-homed
by Keith 21 Jan '14

21 Jan '14

Hi Daniel, So I have looked at the packet capture, the "SIP Request: UPDATE" comes in but when that request is forwarded to the phone it sends it from the interface IP and not the sub interface IP. Therefore the message never gets to the phone and the call is hung up after 30 seconds (due to the failure to respond). Cheers Keith

2 1

FW: FW: Regd. t_suspend() and t_continue()
by Shankar 21 Jan '14

21 Jan '14

Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 21 Jan '14

21 Jan '14

We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason On Tue, Jan 21, 2014 at 10:54 AM, Shankar <shankar.rk(a)plintron.com> wrote: > Hi, > > > > Anyone who had used t_suspend() and t_continue() can share the performance > details? > > > > I tried async module with one sec sleep time. I tried only 5 calls per > second but still it was not successful. After sometime I see below logs, > > > > Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: > t_continue(): ERROR: t_continue: transaction not found > > Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times > > Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times > > Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times > > > > Any configuration changes can help here? > > > > Regards, > > Shankar > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Wednesday, January 15, 2014 1:26 PM > *To:* 'Jason Penton' > > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Jason, > > > > I am using 4.0.2 > > > > Regards, > > Shankar > > > > *From:* Jason Penton [mailto:jason.penton@smilecoms.com<jason.penton(a)smilecoms.com>] > > *Sent:* Wednesday, January 15, 2014 1:21 PM > *To:* Shankar > *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Subject:* Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > > > Hi Shankar, > > > > What version of Kamailio are you running? Kamailio -V > > > > Cheers > > Jason > > > > On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: > > Hi Jason, > > > > Please find below my response inline, > > > > > > I have some questions for you as we have used suspend/continue quite a lot > in the IMS code and don't have any leaks. > > > > Firstly, why are you using pkg_mem for your hash_id and label? Remember > that you will be in 2 different processes in the suspend and continue > portions of the code... so pkg_mem will not work - you should use shm_mem > instead. > > > > [Shankar] We use pkg_mem because we are invoking t_continue from the same > process ( using thread ). > > > > Secondly, how are you using top to tell that you have a leak? Kamailio's > memory is internally managed. > > > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the shared > usage of shared memory for the process. > > > > Cheers > > Jason > > > > > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > > > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com<shankar.rk(a)plintron.com> > ] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > > > > > > -- > > > > *Jason Penton* > > *Senior Manager: Applications and Services* > > *Smile Communications Pty (Ltd)* > > *Mobile:* > > +27 (0) 83 283 7000 > > *Skype:* > > jason.barry.penton > > jason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> > > www.smilecoms.com > > > > This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer > > > > > > > > > _______________________________________________ > 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 > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Daniel-Constantin Mierla 21 Jan '14

21 Jan '14

Hello, if you do: kamctl mi get_statistics all do you see an increase of active transactions counter? Would be possible for you to compile with MEMDBG=1 and then get the memory logs? make MEMDBG=1 cfg Then add your extra modules in include_modules variable inside modules.lst Afterwards: make install Edit kamailio.cfg and set memlog global parameter to a lower value than debug global parameter. Run kamailio for a while with the new config, when you see the shared memory keeps increasing, stop kamailio. Send the log messages from syslog to mailing list or made them available for download somewhere. Cheers, Daniel On 21/01/14 09:54, Shankar wrote: > > Hi, > > Anyone who had used t_suspend() and t_continue() can share the > performance details? > > I tried async module with one sec sleep time. I tried only 5 calls per > second but still it was not successful. After sometime I see below logs, > > Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm > [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found > > Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times > > Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times > > Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times > > Any configuration changes can help here? > > Regards, > > Shankar > > *From:*Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Wednesday, January 15, 2014 1:26 PM > *To:* 'Jason Penton' > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > *Subject:* RE: [SR-Users] FW: Regd. t_suspend() and t_continue() > > Hi Jason, > > I am using 4.0.2 > > Regards, > > Shankar > > *From:*Jason Penton [mailto:jason.penton@smilecoms.com] > *Sent:* Wednesday, January 15, 2014 1:21 PM > *To:* Shankar > *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Subject:* Re: [SR-Users] FW: Regd. t_suspend() and t_continue() > > Hi Shankar, > > What version of Kamailio are you running? Kamailio -V > > Cheers > > Jason > > On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com > <mailto:shankar.rk@plintron.com>> wrote: > > Hi Jason, > > Please find below my response inline, > > I have some questions for you as we have used suspend/continue quite a > lot in the IMS code and don't have any leaks. > > Firstly, why are you using pkg_mem for your hash_id and label? > Remember that you will be in 2 different processes in the suspend and > continue portions of the code... so pkg_mem will not work - you should > use shm_mem instead. > > [Shankar] We use pkg_mem because we are invoking t_continue from the > same process ( using thread ). > > Secondly, how are you using top to tell that you have a leak? > Kamailio's memory is internally managed. > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the > shared usage of shared memory for the process. > > Cheers > > Jason > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com > <mailto:shankar.rk@plintron.com>> wrote: > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command > results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the > same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > > -- > > > *Jason Penton* > > *Senior Manager: Applications and Services* > > *Smile Communications Pty (Ltd)* > > *Mobile:* > > > > +27 (0) 83 283 7000 > > *Skype:* > > > > jason.barry.penton > > jason.penton(a)smilecoms.com <mailto:name.surname@smilecoms.com> > > www.smilecoms.com <http://www.smilecoms.com/> > > This email is subject to the disclaimer of Smile Communications athttp://www.smilecoms.com/disclaimer > > > > > > _______________________________________________ > 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

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 21 Jan '14

21 Jan '14

Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: > Hi Jason, > > > > Please find below my response inline, > > > > > > I have some questions for you as we have used suspend/continue quite a lot > in the IMS code and don't have any leaks. > > > > Firstly, why are you using pkg_mem for your hash_id and label? Remember > that you will be in 2 different processes in the suspend and continue > portions of the code... so pkg_mem will not work - you should use shm_mem > instead. > > > > [Shankar] We use pkg_mem because we are invoking t_continue from the same > process ( using thread ). > > > > Secondly, how are you using top to tell that you have a leak? Kamailio's > memory is internally managed. > > > > [Shankar] After running for say 20minutes, we get out of shared memory > error. Also in top output we observed incremental increase in the shared > usage of shared memory for the process. > > > > Cheers > > Jason > > > > > > On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > > > > > Re-sending without the attachment. > > > > > > > > > > > > *From:* Shankar [mailto:shankar.rk@plintron.com<shankar.rk(a)plintron.com> > ] > > > *Sent:* Monday, January 13, 2014 4:57 PM > > > *To:* 'sr-users(a)lists.sip-router.org' > > > *Subject:* Regd. t_suspend() and t_continue() > > > > > > > > > > > > Hi, > > > > > > > > > > > > We are trying out the t_suspend() and t_continue() in our test setup. > > > We are facing memory leak ( both shm and pkg as per top command results). > > > > > > > > > > > > Please find below the scenario, > > > > > > > > > > > > 1) Do a t_newtran() > > > > > > 2) Allocate pkg memory for hashid and label. > > > > > > 3) Call t_suspend() > > > > > > 4) Do t_continue() when async result is available > > > > > > 5) De-allocate pkg memory reserved for hashid and label > > > > > > 6) Do a t_relay() which forwards the sip message to another sip > node. > > > > > > > > > > > > In the step (6) above, we see t_newtran() allocates one more time > > > shared memory for the same transaction. > > > > > > > > > > > > We tried t_release() after step (4) to release the transaction as > > > t_relay() anyways allocates new shared memory. Nothing helped. > > > > > > > > > > > > Please let me know what are the logs you would require to debug the same. > > > I am attaching syslog for this run. > > > > > > > > > > > > Regards, > > > > > > Shankar > > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

2 2

Kamailio World 2014 - First group of speakers
by Daniel-Constantin Mierla 21 Jan '14

21 Jan '14

Hello, I would like to announce that the details about first group of speakers to present at Kamailio World Conference are now published at: - http://conference.kamailio.com/k02/speakers/ There are several excellent proposals still under review, we expect to fill the schedule very soon, hurry up if you intend to submit one. So far we have a good mixture of topics with a consistent group of new speakers comparing with the past edition. A selection with the titles of the proposals is available at: - http://conference.kamailio.com/k02/schedule/ Many thanks to the sponsors that make the event happening: FhG Fokus, Asipto, Sipwise, Sipgate, ITCenter, NG Voice. If you plan to attend, consider registering as soon as possible to secure your seat - registration form is available at: - http://conference.kamailio.com/k02/registration/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Non-authentication setup of Kamailio
by Carel Burger 20 Jan '14

20 Jan '14

Hi there, I have never used Kamailio before but want to investigate if if will work in my scenario before I invest time to learn it. I am the administrator of a small Wireless ISP. We do not provide SIP channels to our customers, we rather let them choose a service provider of their choice. Currently our customers are using two different providers. Since we are using ADSL with only one static IP we sometimes run into issues at the providers side with one way audio when our clients make a call to another client which is using the same service provider. I assume this is because of NAT. Since the RTP traffic actually leaves our network and then comes back to the other client the quality of the call is not as good as phoning a non client from the network. I want to know if it would be possible to setup Kamailio to keep the internal network calls traffic from leaving the network, and also allow free phone calls on our network. To do this we would require the authentication to take place on the service providers sip server and not on Kamailio. Kamailio would only do routing of the traffic and not worry about authentication. Would this be possible? Regards, Carel Burger

2 2

Development of a Mobile VOIP app connecting to Kamailio Server
by Neville D'Souza 20 Jan '14

20 Jan '14

Hi, I recently got a Kamailio v4.0.0 server up and running with Siremis v4.0.0 on a Debian Wheezy system with mysql as backend database . The Chat, file sharing ,audio and video services work flawlessly even behind nat and the same need to be incorporated in an app. I have configured rtpproxy and tls with self signed certificates. This is basically a test environment and we will install the same kamailio version with commercial certificates on a live hosted server. Server side everything is working. Problem is on the development side. Our company is venturing into VOIP apps for the first time and trying to look at how to go about it. Firstly we would like to start with an android app and then move on to Iphone, Windows, Blackberry and Symbian app. We need to know what is required on server side (any software to be installed or any configuration on kamailio or addition of modules) and client side ( which library and development environment to use) to make it work. Lastly in terms of security what things I need to take care of in the server and app. Best Regards, Neville D'Souza

2 3

kamailio crash after sqlops failures
by Uri Shacked 19 Jan '14

19 Jan '14

Daniel hi, I attached the following file: 1. "bt full" and "log" for each of the 3 servers. 2. The central MySQL slow queries log 3. The var/log/messages from one server. The servers crashed at different times but: 1. The MySQL server was very slow between 16:10:00 till 16:12:00 (approximately). 2. On each server, I am pretty sure, kamailio crashed while doing the data reload from modules like MTREE, HTABLE, DIALPLAN and CARRIERROUTE (all and all round 100,000 rows). Thanks again, Uri On Thu, Jan 16, 2014 at 2:17 AM, Daniel-Constantin Mierla <daniel(a)asipto.com > wrote: > Hello, > > the backtrace shows a crash in tm module, not sqlops. > > Can you tell which of the log files correspond to the instance that > produced the core file from where you took the back trace you attached? > > Can you give the backtraces from all cores? You say there were 3 crashes. > It is important to see if the backtrace is the same everywhere. > > Cheers, > Daniel >

2 2

Re: [SR-Users] S-CSCF configuration brings to seg fault
by Gennaro Sarnataro 17 Jan '14

17 Jan '14

Hi Carsten, I would like to try my scenarios using a kamailio nightly build but i wasn't able to checkout it. Can you indicate me where to find latest builds? 2014/1/15 Gennaro Sarnataro <gen.sarnataro(a)gmail.com> > hi Carsten, > thank for your responsivness!!! > i'm using 4.1 version installed from repository. > how do you fix it?? > > > 2014/1/15 Carsten Bock <carsten(a)ng-voice.com> > >> Salut Gennaro, >> >> what Version of Kamailio are you using? Did you install it from GIT or >> from what repository? >> Looks similar to a bug, i just fixed yesterday... >> I recommend to use at least a 4.1 nightly build. >> >> Thanks, >> Carsten >> >> 2014/1/15 Gennaro Sarnataro <gsarnataro(a)it.tiscali.com>: >> > hello everyone, >> > I'm using some kamailio configurations to perform I-CSCF and S-CSCF. >> > My configurations file let me register and unregister correctly. >> > When i try to perform an invite I've a memory leak as you can see from >> this >> > log: >> > >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: >> received >> >> packet >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: >> received >> >> register >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth >> >> [authorize.c:663]: authenticate(): uri=sip:ims.test.net >> >> nonce=173692b1aea9182c2ac712f915f0144a >> >> response=f1efa37094e4492af1e292043c3d57ea qop=auth nc=00000001 >> >> cnonce=a2f5a827ac94467a39e81584c11e8284 hbody= >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth >> >> [authorize.c:708]: authenticate(): UE said: >> f1efa37094e4492af1e292043c3d57ea >> >> and we expect f1efa37094e4492af1e292043c3d57ea ha1 >> >> 1c31125e2929b3fd6a609bd1a134a631 (REGISTER) >> >> >> >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_registrar_scscf >> >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : >> >> [saa_return_code] - [-2] >> >> >> >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf >> >> [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: >> Failed >> >> finding avp >> >> >> >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf >> >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : >> >> [saa_return_code] - [1] >> >> >> >> Jan 15 11:07:50 localhost kamailio[2932]: WARNING: tm >> [t_lookup.c:1536]: >> >> t_unref(): WARNING: script writer didn't release transaction >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: ERROR: <script>: INVITE >> >> (sip:+393922568123@ims.test.net (10.139.140.249:14437) to >> >> sip:+39666555@ims.test.net, >> >> >> 0gQAAC8WAAACBAAALxYAAM3V1YJ1oc5VxmgXA6gfuMM+33iKMHHLO4FAjeWDRORBfoRu5acqCjeTKNa2pjdRdA--(a)10.139.140.249 >> ) >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: >> received >> >> packet >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: Mobile >> >> Originating Side >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Routing >> Mobile >> >> Originating side >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Adding angle >> >> brackets to P-Asserted-Identity >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: ims_isc [mod.c:236]: >> >> isc_match_filter(): Checking triggers >> >> >> >> Jan 15 11:08:05 localhost kamailio[2940]: : <core> >> [mem/q_malloc.c:454]: >> >> qm_free(): BUG: qm_free: bad pointer 0x7f3dc03597a0 (out of memory >> block!) >> >> called from <core>: parser/parse_param.c: do_free_params(629) - >> aborting >> >> >> >> Jan 15 11:08:05 localhost kamailio[2943]: : <core> [pass_fd.c:293]: >> >> receive_fd(): ERROR: receive_fd: EOF on 90 >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:775]: >> >> handle_sigs(): child process 2940 exited by a signal 6 >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:778]: >> >> handle_sigs(): core was generated >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: <core> [main.c:790]: >> >> handle_sigs(): INFO: terminating due to SIGCHLD >> >> >> >> Jan 15 11:08:05 localhost kamailio[2939]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> >> >> ..... >> >> >> >> >> >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> Jan 15 11:08:05 localhost kamailio[2876]: INFO: <core> [main.c:841]: >> >> sig_usr(): INFO: signal 15 received >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:251]: >> >> cdp_exit(): CDiameterPeer child stopping ... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:428]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Terminating all >> >> children... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2939] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2938] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2937] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2936] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2935] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2934] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2933] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:435]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child >> [2932] >> >> to terminate... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp >> [diameter_peer.c:447]: >> >> diameter_peer_destroy(): destroy_diameter_peer(): All processes >> terminated. >> >> Cleaning up. >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:137]: >> >> worker_destroy(): Unlocking workers waiting on empty queue... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:140]: >> >> worker_destroy(): Unlocking workers waiting on full queue... >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: CRITICAL: cdp >> >> [diameter_peer.c:487]: diameter_peer_destroy(): >> destroy_diameter_peer(): Bye >> >> Bye from C Diameter Peer test >> >> >> >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:253]: >> >> cdp_exit(): ... CDiameterPeer child stoped >> > >> > >> > after that the S-CSCF terminate. >> > it's unuseful to attach any wireshark dump because,as I told before, the >> > Serving process terminates just after receiving the INVITE. >> > Instead I attach my configuration. >> > >> > thanks in advance!!! >> > >> > saluti >> > >> > Gennaro Sarnataro >> > >> > _______________________________________________ >> > 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 >> > >> >> >> >> -- >> 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/ >> >> _______________________________________________ >> 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 >> > > > > -- > Gennaro Sarnataro > -- Gennaro Sarnataro

2 1

Re: [SR-Users] S-CSCF configuration brings to seg fault
by Gennaro Sarnataro 17 Jan '14

17 Jan '14

hi Carsten, thank for your responsivness!!! how do you fix it?? i'm using 4.1 version installed from repository. I would like to try my scenarios using a kamailio nightly build but i wasn't able to checkout it. Can you indicate me where to find latest builds? 2014/1/15 Carsten Bock <carsten(a)ng-voice.com> > Salut Gennaro, > > what Version of Kamailio are you using? Did you install it from GIT or > from what repository? > Looks similar to a bug, i just fixed yesterday... > I recommend to use at least a 4.1 nightly build. > > Thanks, > Carsten > > 2014/1/15 Gennaro Sarnataro <gsarnataro(a)it.tiscali.com>: > > hello everyone, > > I'm using some kamailio configurations to perform I-CSCF and S-CSCF. > > My configurations file let me register and unregister correctly. > > When i try to perform an invite I've a memory leak as you can see from > this > > log: > > > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: > received > >> packet > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: > received > >> register > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth > >> [authorize.c:663]: authenticate(): uri=sip:ims.test.net > >> nonce=173692b1aea9182c2ac712f915f0144a > >> response=f1efa37094e4492af1e292043c3d57ea qop=auth nc=00000001 > >> cnonce=a2f5a827ac94467a39e81584c11e8284 hbody= > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth > >> [authorize.c:708]: authenticate(): UE said: > f1efa37094e4492af1e292043c3d57ea > >> and we expect f1efa37094e4492af1e292043c3d57ea ha1 > >> 1c31125e2929b3fd6a609bd1a134a631 (REGISTER) > >> > >> Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_registrar_scscf > >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : > >> [saa_return_code] - [-2] > >> > >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf > >> [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: > Failed > >> finding avp > >> > >> Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf > >> [cxdx_sar.c:79]: create_return_code(): created AVP successfully : > >> [saa_return_code] - [1] > >> > >> Jan 15 11:07:50 localhost kamailio[2932]: WARNING: tm [t_lookup.c:1536]: > >> t_unref(): WARNING: script writer didn't release transaction > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: ERROR: <script>: INVITE > >> (sip:+393922568123@ims.test.net (10.139.140.249:14437) to > >> sip:+39666555@ims.test.net, > >> > 0gQAAC8WAAACBAAALxYAAM3V1YJ1oc5VxmgXA6gfuMM+33iKMHHLO4FAjeWDRORBfoRu5acqCjeTKNa2pjdRdA--(a)10.139.140.249 > ) > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: > received > >> packet > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: Mobile > >> Originating Side > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Routing Mobile > >> Originating side > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Adding angle > >> brackets to P-Asserted-Identity > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: INFO: ims_isc [mod.c:236]: > >> isc_match_filter(): Checking triggers > >> > >> Jan 15 11:08:05 localhost kamailio[2940]: : <core> [mem/q_malloc.c:454]: > >> qm_free(): BUG: qm_free: bad pointer 0x7f3dc03597a0 (out of memory > block!) > >> called from <core>: parser/parse_param.c: do_free_params(629) - aborting > >> > >> Jan 15 11:08:05 localhost kamailio[2943]: : <core> [pass_fd.c:293]: > >> receive_fd(): ERROR: receive_fd: EOF on 90 > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:775]: > >> handle_sigs(): child process 2940 exited by a signal 6 > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:778]: > >> handle_sigs(): core was generated > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: <core> [main.c:790]: > >> handle_sigs(): INFO: terminating due to SIGCHLD > >> > >> Jan 15 11:08:05 localhost kamailio[2939]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> > >> ..... > >> > >> > >> Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> Jan 15 11:08:05 localhost kamailio[2876]: INFO: <core> [main.c:841]: > >> sig_usr(): INFO: signal 15 received > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:251]: > >> cdp_exit(): CDiameterPeer child stopping ... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:428]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Terminating all > >> children... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2939] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2938] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2937] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2936] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2935] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2934] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2933] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:435]: > >> diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child > [2932] > >> to terminate... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp > [diameter_peer.c:447]: > >> diameter_peer_destroy(): destroy_diameter_peer(): All processes > terminated. > >> Cleaning up. > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:137]: > >> worker_destroy(): Unlocking workers waiting on empty queue... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:140]: > >> worker_destroy(): Unlocking workers waiting on full queue... > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: CRITICAL: cdp > >> [diameter_peer.c:487]: diameter_peer_destroy(): > destroy_diameter_peer(): Bye > >> Bye from C Diameter Peer test > >> > >> Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:253]: > >> cdp_exit(): ... CDiameterPeer child stoped > > > > > > after that the S-CSCF terminate. > > it's unuseful to attach any wireshark dump because,as I told before, the > > Serving process terminates just after receiving the INVITE. > > Instead I attach my configuration. > > > > thanks in advance!!! > > > > saluti > > > > Gennaro Sarnataro > > > > _______________________________________________ > > 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 > > > > > > -- > 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/ > > _______________________________________________ > 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

kamailio-4.0.2 core issue
by Surendra 17 Jan '14

17 Jan '14

Hi, We are facing a core issue in 4.0.2 version. Please find the GDB trace below #0 udp_rcv_loop () at udp_server.c:471 471 fromlen=sockaddru_len(bind_address->su); (gdb) bt #0 udp_rcv_loop () at udp_server.c:471 #1 0x0000000000475c14 in main_loop () at main.c:1851 #2 0x000000000047a98b in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2939 We have simulated this so many times in load mode. Machine details: Linux PLT-RA-RD-W167A.localdomain 2.6.18-308.el5 #1 SMP Fri Jan 27 17:17:51 EST 2012 x86_64 x86_64 x86_64 GNU/Linux Can anyone help me on this Regards Surendra

1 0

kamailio-4.1.1 avp issue within failure routes
by José Seabra 17 Jan '14

17 Jan '14

Hi there, I upgraded my kamailio server from version 4.0.1 to 4.1.1, then i noticed that all my "AVP's variavels" that are initialized on initial "INVITE", are all set as null when call goes to failure route, so kamailio is not keeping the variavel values for same call(dialog) When it happens i see the following warning "WARNING: tm [t_lookup.c:1536]: t_unref(): WARNING: script writer didn't release transaction" Thanks for your help -- Regards José Seabra

3 11

Memory Leaks with Kamailio using SEAS module
by ziad habchi 17 Jan '14

17 Jan '14

Dear Support, I am having an "out of memory" problem while using kamailio with SEAS module. After using Kamailio with the SEAS module for few hours with a high load , I am getting "out of memory" error that I believe is caused from a memory leak. I would like your help interpreting this error. The output of kamailio -v is : version: kamailio 4.0.4 (i386/linux) cabe58 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, 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: cabe58 compiled on 14:23:27 Dec 2 2013 with gcc 4.1.2 kamctl ps : Process:: ID=0 PID=31191 Type=attendant Process:: ID=1 PID=31192 Type=udp receiver child=0 sock=193.100.200.18:5070 Process:: ID=2 PID=31193 Type=udp receiver child=1 sock=193.100.200.18:5070 Process:: ID=3 PID=31194 Type=udp receiver child=2 sock=193.100.200.18:5070 Process:: ID=4 PID=31195 Type=udp receiver child=3 sock=193.100.200.18:5070 Process:: ID=5 PID=31196 Type=slow timer Process:: ID=6 PID=31197 Type=timer Process:: ID=7 PID=31198 Type=ctl handler Process:: ID=8 PID=31199 Type=MI FIFO Process:: ID=9 PID=31200 Type=SEAS Process:: ID=10 PID=31201 Type=tcp receiver (generic) child=0 Process:: ID=11 PID=31202 Type=tcp receiver (generic) child=1 Process:: ID=12 PID=31203 Type=tcp receiver (generic) child=2 Process:: ID=13 PID=31204 Type=tcp receiver (generic) child=3 Process:: ID=14 PID=31205 Type=tcp main process kamcmd pkg.stats: { entry: 0 pid: 31191 rank: 0 used: 72480 free: 4091840 real_used: 102448 } { entry: 1 pid: 31192 rank: 1 used: 80624 free: 4083696 real_used: 110592 } { entry: 2 pid: 31193 rank: 2 used: 80624 free: 4083696 real_used: 110592 } { entry: 3 pid: 31194 rank: 3 used: 80624 free: 4083696 real_used: 110592 } { entry: 4 pid: 31195 rank: 4 used: 80624 free: 4083696 real_used: 110592 } { entry: 5 pid: 31196 rank: -1 used: 2276560 free: 4083744 real_used: 17770256 } { entry: 6 pid: 31197 rank: -1 used: 1787296 free: 4083744 real_used: 17280992 } { entry: 7 pid: 31198 rank: -2 used: 77760 free: 4086544 real_used: 107744 } { entry: 8 pid: 0 rank: 0 used: 89920 free: 4074352 real_used: 119936 } { entry: 9 pid: 0 rank: 0 used: 0 free: 0 real_used: 0 } { entry: 10 pid: 31201 rank: 5 used: 140240 free: 4023728 real_used: 170560 } { entry: 11 pid: 31202 rank: 6 used: 140240 free: 4023712 real_used: 170576 } { entry: 12 pid: 31203 rank: 7 used: 140240 free: 4023856 real_used: 170432 } { entry: 13 pid: 31204 rank: 8 used: 140240 free: 4023712 real_used: 170576 } { entry: 14 pid: 31205 rank: -4 used: 3877344 free: 4030176 real_used: 19269056 } As you can see above , the output of the SEAS module is not showing (pid 31200) Moreover , I can see that the real_used value in the kamcmd core.shmmem keep on increasing , and free decrease until I finally get an error "out of memory" When my SEAS client exit the output of kamcmd core.shmmem : { total: 33554432 free: 16273440 used: 1787296 real_used: 17280992 max_used: 33553520 fragments: 8098 } The traffic is a simple SIP MESSAGE from a seagull simulator, with a 200OK reply from my SEAS client. I appreciate your support, Thank you in advance, Ziad Habchi

3 10

Help required
by Gerhard De Villiers-Mohr 17 Jan '14

17 Jan '14

Good day Can you please forward me a manual of some sort or assist with setting up the kamailio serimis 4.0 to accept registration of IP PBX systems or soft phones. I am very new to this and need a step by step instructions. Also I will be getting a sip trunk to use for testing purposes before we go live and need help with setting this up and routing calls from users to the sip trunk for outbound calls. Please assist me , this is very new to me. If you prefer to do a remote login session I will gladly provide you with the teamviewer session details. Sincerely. Description: Description: Description: gerhard mohr email signature

1 0

Fosdem this year
by Daniel-Constantin Mierla 16 Jan '14

16 Jan '14

Hello, not much about real time communications this year at Fosdem, but being a convenient venue for me, the plan at this moment is to go . I saw some notes online from various people in the community that planned to attend as well, maybe we can synchronize and have an informal meeting around and/or a dinner. Who is planning to go? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

9 13

Kamailio multi domain issue..
by arun Jayaprakash 16 Jan '14

16 Jan '14

Hello, I am using Kamailio with a multidomain set up. It is working well but for one issue: I first populate the /etc/hosts file with the domain name and the IP address ( and of course the domain table ) but the new domain does not take effect till I restart the kamailio. Does anyone know how I can overcome this problem as it is not possible for me to keep restarting the server each time a new user registers. Thanks for the help. Arun

2 3

Kamailio Advanced Training Course in Europe 2014?
by Yufei Tao 16 Jan '14

16 Jan '14

Hi Just a quick question about the Kamailio Advanced Training Course: are there any courses scheduled in Europe in 2014? If there are, roughly what time will they likely to take place? The only one I can find is the one in Miami next month but it's a bit far to travel. Cheers, 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

1 0

Authenticate Failure
by Keith 16 Jan '14

16 Jan '14

Hi, Having a strange issue: auth_db [authorize.c:476]: auth_check(): realm [x.x.x.x] table [subscriber] flags [1] Sep 8 02:10:28 localhost /usr/sbin/kamailio[65448]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm 'x.x.x.x' not found Sep 8 02:10:28 localhost /usr/sbin/kamailio[65448]: DEBUG: auth_db [authorize.c:252]: digest_authenticate(): no credentials The x is replaced with my IP. The user was created with the IP address as the domain. Username and password are correct and the phone registers fine. It's just when I place a call it doesn't get routed properly. Thanks Keith

2 1

kamailio crash after sqlops failures
by Uri Shacked 16 Jan '14

16 Jan '14

Hi, Our system had crashed again. Here are the exact details and there are logs and corefile attached. We use 6 kamailio servers over 3 sites. All the servers are connecting to 1 DB server running MYSQL 5.6.12. The sql server supplies 2 things for kamailio servers. 1. Some SQLOPS that the servers need. 2. The data for the MTREE, HTABLE, CARRIERROUTE etc.. We use "kamctl" command in shell script to reload the data for the modules every 1 minute. In normal times, the reload takes about 2-3 sec each minute. When the sql server is down, the queries and reloads fail and it is OK. Yesterday, we had a different failure on the sql server – The server is set ona centos 6.2 that is under a VM cluster. The VM cluster had some disk writing errors, and the sql server could not write or read fast enough. On 3 of the 6 servers (one at each site) kamailio crashed. I have no idea why. When the sql server is down, it does not happen. I attached all 6 servers log, 1 server core file, 1 server var/log/messages and the mysql slow-query log. Any ideas how to handle this? kamailio version 3.3.2 BR, Uri

2 1

Daily crash of kamailio 4.0.5
by Tristan Mahé 15 Jan '14

15 Jan '14

Hi guys, I'm having some crashs on a simple REGISTRAR, in st_delete_ucontact. It happens daily, mostly when proxy is loaded. Saw that someone got the same problem via this pastebin: http://pastebin.com/GsMtav7A. Does someone got an idea of why it's failing or a workaround or a fix to share ? Here is the backtrace for us: #0 0x00007fb6d7178e3c in st_delete_ucontact (_c=0xf0) at ucontact.c:358 #1 0x00007fb6d718f4a1 in delete_ucontact (_r=0x7fb69c9a1ea0, _c=0xf0) at urecord.c:509 #2 0x00007fb6d6d49b6a in update_contacts (_m=0x7fb6d8e1ab70, _r=0x7fb69c9a1ea0, _mode=1) at save.c:700 #3 0x00007fb6d6d4a4c3 in add_contacts (_m=0x7fb6d8e1ab70, _d=0x7fb6908c2f10, _a=0x7fff02a69380, _mode=1) at save.c:803 #4 0x00007fb6d6d4a9f2 in save (_m=0x7fb6d8e1ab70, _d=0x7fb6908c2f10, _cflags=4, _uri=0x0) at save.c:879 #5 0x00007fb6d6d3e41d in w_save2 (_m=0x7fb6d8e1ab70, _d=0x7fb6908c2f10 "\240.\214\220\266\177", _cflags=0x4 <Address 0x4 out of bounds>) at reg_mod.c:442 #6 0x000000000041b80b in do_action (h=0x7fff02a699a0, a=0x7fb6d8de2a30, msg=0x7fb6d8e1ab70) at action.c:1086 #7 0x0000000000423f62 in run_actions (h=0x7fff02a699a0, a=0x7fb6d8de2a30, msg=0x7fb6d8e1ab70) at action.c:1573 #8 0x0000000000424639 in run_actions_safe (h=0x7fff02a6ae90, a=0x7fb6d8de2a30, msg=0x7fb6d8e1ab70) at action.c:1637 #9 0x00000000004bef7d in rval_get_int (h=0x7fff02a6ae90, msg=0x7fb6d8e1ab70, i=0x7fff02a69da8, rv=0x7fb6d8de37a8, cache=0x0) at rvalue.c:920 #10 0x00000000004c1b85 in rval_expr_eval_int (h=0x7fff02a6ae90, msg=0x7fb6d8e1ab70, res=0x7fff02a69da8, rve=0x7fb6d8de37a0) at rvalue.c:1914 #11 0x00000000004c1d6c in rval_expr_eval_int (h=0x7fff02a6ae90, msg=0x7fb6d8e1ab70, res=0x7fff02a6a200, rve=0x7fb6d8de30c0) at rvalue.c:1922 #12 0x000000000041b422 in do_action (h=0x7fff02a6ae90, a=0x7fb6d8df3700, msg=0x7fb6d8e1ab70) at action.c:1050 #13 0x0000000000423f62 in run_actions (h=0x7fff02a6ae90, a=0x7fb6d8de27f0, msg=0x7fb6d8e1ab70) at action.c:1573 #14 0x000000000041b661 in do_action (h=0x7fff02a6ae90, a=0x7fb6d8df3900, msg=0x7fb6d8e1ab70) at action.c:1065 #15 0x0000000000423f62 in run_actions (h=0x7fff02a6ae90, a=0x7fb6d8df3900, msg=0x7fb6d8e1ab70) at action.c:1573 #16 0x0000000000419851 in do_action (h=0x7fff02a6ae90, a=0x7fb6d8dc0e70, msg=0x7fb6d8e1ab70) at action.c:690 #17 0x0000000000423f62 in run_actions (h=0x7fff02a6ae90, a=0x7fb6d8db9bf0, msg=0x7fb6d8e1ab70) at action.c:1573 #18 0x0000000000424701 in run_top_route (a=0x7fb6d8db9bf0, msg=0x7fb6d8e1ab70, c=0x0) at action.c:1658 #19 0x00000000004a0018 in receive_msg ( buf=0x9055c0 "REGISTER sip:[ADOMAIN] SIP/2.0\r\nVia: SIP/2.0/UDP [ANIP];branch=z9hG4bKcydzigwkX;i=20be2b3\r\nVia: SIP/2.0/TLS [ANIP]:40080;received=[ANIP];rport=50419;branch=z9hG4bK793894041\r\nFr"..., len=613, rcv_info=0x7fff02a6b140) at receive.c:211 #20 0x0000000000530665 in udp_rcv_loop () at udp_server.c:557 #21 0x0000000000469be4 in main_loop () at main.c:1638 #22 0x000000000046ca85 in main (argc=13, argv=0x7fff02a6b478) at main.c:2566 And here is the simple part of config that could call this method: route[REGISTRAR] { if (is_method("REGISTER")) { if (!save("location","0x04")) { sl_reply_error(); } else { # DOSOMESTUFF like replicate to a backup proxy } exit; } } If you need more info, don't hesitate to ask !!

1 0

Link Redundancy
by Owais ul Haq 15 Jan '14

15 Jan '14

Hello, I have a kamailio server deployed along with an internet connection and database (on a seperate server than the kamailio server) obviously. Problem in my scenario is that my internet connection is not comfortably reliable and in my environment I just cant help myself in this regard. I want to get another internet connection with another public ip, attach another kamailio server (of same configurations as the first one) with it and then use the same database for both kamailio servers (join both kamailio servers to the same database server). So if one of my connections is down, calls can use other connection attached to other server (take the alternate path) and initialize through the second server. Now my questions are : a) How can I assign two ip's against one domain name ?? b) How can calls be routed to second IP if kamailio server attached to first IP is not responding ?? c) Is all this really possible ?? Or just a fruit less effort of my mind :( Cheers.

2 1

Re: [SR-Users] register fails from openvpn clients
by meres 15 Jan '14

15 Jan '14

Hello, Thank you for your help The REGISTER request is getting to kamailio. I attach my tcpdump output: http://pastebin.com/3RV9wG5G Regards, Kostas On Jan 15, 2014, at 11:09 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > what do you mean by "register fails"? > > Is the REGISTER request getting to kamailio? Is there a reply for it? > > A ngrep output taken on kamailio server for such registration would help to figure out where is the issue. > > Cheers, > Daniel > > On 13/01/14 18:08, meres wrote: >> I have kamailio 4.1.0 running on a server on a real ip and behind a firewall. >> NAT detection is enabled on kamailio because many remote clients are behind NAT, so NAT is working fine along with rtpproxy. >> Everything else (incoming, outgoing) is working fine except the following: >> >> Users who are connected to our openvpn server (bridged mode) which is on the same subnet with kamailio, fail to register. >> I suspect that kamailio detects NAT on these clients as all of them are behind NAT, but they also have obtained a real ip from our openvpn server on their tap interface and as a result, REGISTER fails. >> >> One solution, but not the best is to exclude kamailio from our openvpn routes but I would not prefer this because I wil not be able to manage the server remotely >> >> My config: http://pastebin.com/JSxzgmKH >> >> Any suggestions? >> >> >> >> >> _______________________________________________ >> 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 > _______________________________________________ > 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

3 5

Implementing the Presence module(Kamailio) to work with websockets (sipML5)
by אורן אברהם 15 Jan '14

15 Jan '14

Dear List, I have some strange error when using sipML5 client (over websockets) with Kamailio 4.0 server when sending a SUBSCRIBE request. here is my log print: Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: WARNING: <core> [msg_translator.c:2499]: TCP/TLS connection (id: 0) for WebSocket could not be found (tcp:10.0.0.4:8080) Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: tm [t_msgbuilder.c:1367]: assemble_via: via building failed Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: tm [t_msgbuilder.c:1540]: build_uac_req(): Error while assembling Via Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: tm [uac.c:338]: t_uac: Error while building message Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [notify.c:1591]: in function tmb.t_request_within Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [notify.c:1678]: sending Notify not successful Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [subscribe.c:678]: Could not send notify Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [subscribe.c:713]: occured Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [subscribe.c:994]: in update_subscription Could anyone help here ? I think it is only some configuration problem, but i'm not sure how to solve it.

3 6

dispatcher sets - mixed/overlapping algorithm
by Rob Moore 15 Jan '14

15 Jan '14

Hi All, I've been trying to find the best way of providing a single dispatcher set to cover all of our outbound SIPtoPSTN providers. However in order to create a single dispatcher set I would need to have a mixture of the available algorithms to provide the required effect. So for example, if I have 4 providers, the first 2 have 2 IP addresses each and the other 2 only have a IP address each. As a standard I would use dispatcher algorithm "8 - use first destination" to move through the 4 providers IP address, 1 at a time as we have a preference of which provider we would like to be used first. However due to CPS and other throughput reasons we get better performance from our first two providers if we round-robin our requests over the two IP addresses for each provider. So What I'm trying to create is a list where I can : round-robin the 2 IPs from Provider1 if both Provider 1 IPs fail I can then.... round-robin the 2 IPs from Provider2 if both Provider 2 IPs fail I can then.... Dispatch calls to the single IP of Provider 3 If Provider 3 fails then.... Dispatch calls to the single IP of Provider 4 I'm, fairly certain that there isn't anything available straight out of the module for this but I'm not sure on how best to build this functionality in. Maybe I need to look at using multiple sets but hidden behind a single set ID. My main preference is that I want to keep all of the logic for this within Kamailio and have our telephony app servers simply dispatcher a call with a specified dispatcher ID as per usual. Other than this one caveat I'm really open to suggestions on how to fulfil this requirement. Any assistance will be greatly appreciated. Many Thanks Rob

1 0

S-CSCF configuration brings to seg fault
by Gennaro Sarnataro 15 Jan '14

15 Jan '14

hello everyone, I'm using some kamailio configurations to perform I-CSCF and S-CSCF. My configurations file let me register and unregister correctly. When i try to perform an invite I've a memory leak as you can see from this log: Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: received > packet Jan 15 11:07:50 localhost kamailio[2873]: INFO: <script>: route: received > register Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth [authorize.c:663]: > authenticate(): uri=sip:ims.test.netnonce=173692b1aea9182c2ac712f915f0144a > response=f1efa37094e4492af1e292043c3d57ea qop=auth nc=00000001 > cnonce=a2f5a827ac94467a39e81584c11e8284 hbody= Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_auth [authorize.c:708]: > authenticate(): UE said: f1efa37094e4492af1e292043c3d57ea and we expect > f1efa37094e4492af1e292043c3d57ea ha1 1c31125e2929b3fd6a609bd1a134a631 > (REGISTER) Jan 15 11:07:50 localhost kamailio[2873]: INFO: ims_registrar_scscf > [cxdx_sar.c:79]: create_return_code(): created AVP successfully : > [saa_return_code] - [-2] Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf > [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed > finding avp Jan 15 11:07:50 localhost kamailio[2932]: INFO: ims_registrar_scscf > [cxdx_sar.c:79]: create_return_code(): created AVP successfully : > [saa_return_code] - [1] Jan 15 11:07:50 localhost kamailio[2932]: WARNING: tm [t_lookup.c:1536]: > t_unref(): WARNING: script writer didn't release transaction Jan 15 11:08:05 localhost kamailio[2940]: ERROR: <script>: INVITE ( > sip:+393922568123@ims.test.net (10.139.140.249:14437) to > sip:+39666555@ims.test.net, > 0gQAAC8WAAACBAAALxYAAM3V1YJ1oc5VxmgXA6gfuMM+33iKMHHLO4FAjeWDRORBfoRu5acqCjeTKNa2pjdRdA--(a)10.139.140.249 > ) Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: received > packet Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: route: Mobile > Originating Side Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Routing Mobile > Originating side Jan 15 11:08:05 localhost kamailio[2940]: INFO: <script>: Adding angle > brackets to P-Asserted-Identity Jan 15 11:08:05 localhost kamailio[2940]: INFO: ims_isc [mod.c:236]: > isc_match_filter(): Checking triggers Jan 15 11:08:05 localhost kamailio[2940]: : <core> [mem/q_malloc.c:454]: > qm_free(): BUG: qm_free: bad pointer 0x7f3dc03597a0 (out of memory block!) > called from <core>: parser/parse_param.c: do_free_params(629) - aborting Jan 15 11:08:05 localhost kamailio[2943]: : <core> [pass_fd.c:293]: > receive_fd(): ERROR: receive_fd: EOF on 90 Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:775]: > handle_sigs(): child process 2940 exited by a signal 6 Jan 15 11:08:05 localhost kamailio[2863]: ALERT: <core> [main.c:778]: > handle_sigs(): core was generated Jan 15 11:08:05 localhost kamailio[2863]: INFO: <core> [main.c:790]: > handle_sigs(): INFO: terminating due to SIGCHLD Jan 15 11:08:05 localhost kamailio[2939]: INFO: <core> [main.c:841]: > sig_usr(): INFO: signal 15 received Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: > sig_usr(): INFO: signal 15 received > ..... > Jan 15 11:08:05 localhost kamailio[2942]: INFO: <core> [main.c:841]: > sig_usr(): INFO: signal 15 received Jan 15 11:08:05 localhost kamailio[2876]: INFO: <core> [main.c:841]: > sig_usr(): INFO: signal 15 received Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:251]: > cdp_exit(): CDiameterPeer child stopping ... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:428]: > diameter_peer_destroy(): destroy_diameter_peer(): Terminating all > children... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2939] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2938] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2937] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2936] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2935] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2934] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2933] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:435]: > diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2932] > to terminate... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [diameter_peer.c:447]: > diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated. > Cleaning up. Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:137]: > worker_destroy(): Unlocking workers waiting on empty queue... Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [worker.c:140]: > worker_destroy(): Unlocking workers waiting on full queue... Jan 15 11:08:05 localhost kamailio[2863]: CRITICAL: cdp > [diameter_peer.c:487]: diameter_peer_destroy(): destroy_diameter_peer(): > Bye Bye from C Diameter Peer test Jan 15 11:08:05 localhost kamailio[2863]: INFO: cdp [mod.c:253]: > cdp_exit(): ... CDiameterPeer child stoped after that the S-CSCF terminate. it's unuseful to attach any wireshark dump because,as I told before, the Serving process terminates just after receiving the INVITE. Instead I attach my configuration. thanks in advance!!! saluti Gennaro Sarnataro

2 1

SIP OPTIONS
by Keith 15 Jan '14

15 Jan '14

Hi, I have a trunk trying to connect to Kamailio but when it sends me OPTIONS I am not replying with 200 OK. This trunk is no authenticated in any way. Any ideas? Keith

3 2

register fails from openvpn clients
by meres 15 Jan '14

15 Jan '14

I have kamailio 4.1.0 running on a server on a real ip and behind a firewall. NAT detection is enabled on kamailio because many remote clients are behind NAT, so NAT is working fine along with rtpproxy. Everything else (incoming, outgoing) is working fine except the following: Users who are connected to our openvpn server (bridged mode) which is on the same subnet with kamailio, fail to register. I suspect that kamailio detects NAT on these clients as all of them are behind NAT, but they also have obtained a real ip from our openvpn server on their tap interface and as a result, REGISTER fails. One solution, but not the best is to exclude kamailio from our openvpn routes but I would not prefer this because I wil not be able to manage the server remotely My config: http://pastebin.com/JSxzgmKH Any suggestions?

2 1

The RPC function in the TM module
by 无愿无悔 15 Jan '14

15 Jan '14

Hello everyone: I want to know what is the parameter for the RPC function "tm.cancel" in the TM module ? I input the parameters such as : tm.cancel d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ 2 but it can accept the only parameter,and inform me "error 400-Callid and CSeq expected as parameters " . The context of the Callid.s is "d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ". But the context of the CSeq.s is "if you get this string, you don't check rpc_scan return code!!!(very bad)". Could you give me an example ? Message: 3 Date: Mon, 13 Jan 2014 09:00:53 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] The RPC Function in The TM Module Message-ID: <52D39D35.5000205(a)gmail.com> Content-Type: text/plain; charset="utf-8"; Format="flowed" Hello, can you try to put callid in between double quotes. Try with the number only and if it doesn't work, try also with number and method. Like: kamcmd tm.cancel "CALLID" "NNN" kamcmd tm.cancel "CALLID" "NNN INVITE" With kamcmd should be an option to force string input parameter to rpc command, iirc, like: kamcmd tm.cancel "CALLID" s:NNN Let us know if it works. Cheers, Daniel hi, I try to put callid between double quotes.Like: tm.cancel "CALLID" "NNN" but it inform : error: 400-Transactions not found. the context of the callid is " "CALLID" " the context of the Cseq is " "NNN" " . note: Thus,the double quotes is treated as a part of the string.

2 1

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Shankar 15 Jan '14

15 Jan '14

Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar

1 0

is_in_subnet & Multiple CIDR subnets
by jay binks 15 Jan '14

15 Jan '14

So Im looking at a way of implementing IP Network ACL's in kamailio.. block all except specific subnets etc.. it seems I can do what I need with the IPOps module and is_in_subnet but is_in_subnet will only take 1 CIDR notation subnet, and I want to be able to put ( For example ) 192.168.1.1/24,172.16.10.1/24 in there ... ( or any number of subnets really ) is there a benevolent kamailio developer on the list who is able to add this simple feature for me ? -- Sincerely Jay

3 3

Re: [SR-Users] Update R-URI after lookup("location")
by Igor Potjevlesch 14 Jan '14

14 Jan '14

Hello, No one has an idea? I was thinking that each request goes to RELAY but even if I try to modify the R-URI in this route, it fails. I still had the contact URI taken from location instead of my modification. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : lundi 13 janvier 2014 17:15 À : sr-users(a)lists.sip-router.org Objet : Update R-URI after lookup("location") Hello, I do an association between X aliases with 1 contact. This contact is connected one time in the location. But sometimes, for some scenario, this contact can be connected 2 or 3 times and the INVITE are sent with parallel forking. When an INVITE is received, after lookup(aliases), I set the R-URI with the original SIP TO with the following instruction: avp_pushto("$ruri/username", "$tU"); before relay. The issue occurred with the next contacts. They dont passed through this instruction. How can I update all available R-URI in location before relay? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 2

Registration to Kamailio World 2014 is open
by Daniel-Constantin Mierla 14 Jan '14

14 Jan '14

Hello, I would like to announce that the registration to Kamailio World 2014 is now open. This year there is also a pre-conference event for half a day on April 2, with technical tutorials for those looking on more practical and hands-on content. You can see more details on the web site at: - http://conference.kamailio.com/k02/registration/ - http://conference.kamailio.com/k02/2014/01/registration-is-open/ Looking forward to meeting many of you in Berlin during April 2-4, 2014. Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

Help with authenticating using Kamailio
by Kasinath 14 Jan '14

14 Jan '14

Hi All, I just installed Kamailio in one server and Asterisk in another. Asterisk loads it sipusers info from database which is in Kamailio server. I don't know how to go further. How can I authenticate Asterisk users through Kamailio. I am trying to authenticate using a sipphone. But no luck. I am missing alot here. I know we can add users using the following command >> kamctl add username password But I already have users in asterisk realtime db. Is there any difference? Awaiting your reply, Thanks in advance, -- Thanks & Regards, Kasinath K P System Engineer Admin-Ahead Server Technologies email:kasinathk@admin-ahead.com

2 1

WebSocket with MSRP
by mohammed alyaseen 14 Jan '14

14 Jan '14

Hi All, I am using Kamailio 4.1 as SIP proxy and registrar with websocket and msrp modules. As a client I am using the JsSIP stack. The client can open websocket for SIP messages by calling "new WebSocket('ws://1.2.3.4:8000','sip')" with sip as a subprotocol, this works fine. But when the client tries to open a websocket for MSRP sessions using new WebSocket('ws://1.2.3.4:8000','msrp') in the browser log I see the error 'WebSocket connection to 'ws://1.2.3.4:8000/' failed: Unexpected response code: 400' . For MSRP file transfer I am using the crocodile msrp lib. The HTTP GET request of sent by the client is: GET / HTTP/1.1 Via: SIP/2.0/TCP 10.147.66.197:54818 Upgrade: websocket Connection: Upgrade Host: 192.168.144.48:8000 Origin: http://localhost Sec-WebSocket-Protocol: msrp Pragma: no-cache Cache-Control: no-cache Sec-WebSocket-Key: gccXstTCf+egnPtufU2xng== Sec-WebSocket-Version: 13 Sec-WebSocket-Extensions: x-webkit-deflate-frame User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/31.0.1650.63 Safari/537.36 Cookie: PHPSESSID=pdlbnc3h24i5r8rdg1979tino2 In the log of Kamailio I see these lines: 11(2114) WARNING: websocket [ws_handshake.c:318]: ws_handle_handshake(): required headers not present 11(2114) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 11(2114) DEBUG: <core> [msg_translator.c:204]: check_via_address(): check_via_address(10.147.66.197, 10.147.66.197, 0) 11(2114) DEBUG: <core> [tcp_main.c:2320]: tcpconn_send_put(): tcp_send: send from reader (2114 (11)), reusing fd 11(2114) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 11(2114) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f37ab162bd0 n=182 fd=8 11(2114) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= HTTP/1.1 400 Bad Request Via: SIP/2.0/TCP 10.147.66.197:54744 Sec-WebSocket-Protocol: sip Sec-WebSocket-Version: 13 Server: kamailio (4.1.1 (x86_64/linux)) Content-Length: 0 So, anyone has an idea whats the problem is? Thanks, Medo

1 0

Update R-URI after lookup("location")
by Igor Potjevlesch 13 Jan '14

13 Jan '14

Hello, I do an association between X aliases with 1 contact. This contact is connected one time in the location. But sometimes, for some scenario, this contact can be connected 2 or 3 times and the INVITE are sent with parallel forking. When an INVITE is received, after lookup("aliases"), I set the R-URI with the original SIP TO with the following instruction: avp_pushto("$ruri/username", "$tU"); before relay. The issue occurred with the next contacts. They don't passed through this instruction. How can I update all available R-URI in location before relay? Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Jason Penton 13 Jan '14

13 Jan '14

Hi, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. We > are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time shared > memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar > > > > > > _______________________________________________ > 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 > > -- *Jason Penton**Senior Manager: Applications and Services**Smile Communications Pty (Ltd)**Mobile:*+27 (0) 83 283 7000*Skype:* jason.barry.pentonjason.penton(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

FW: Regd. t_suspend() and t_continue()
by Shankar 13 Jan '14

13 Jan '14

Re-sending without the attachment. From: Shankar [mailto:shankar.rk@plintron.com] Sent: Monday, January 13, 2014 4:57 PM To: 'sr-users(a)lists.sip-router.org' Subject: Regd. t_suspend() and t_continue() Hi, We are trying out the t_suspend() and t_continue() in our test setup. We are facing memory leak ( both shm and pkg as per top command results). Please find below the scenario, 1) Do a t_newtran() 2) Allocate pkg memory for hashid and label. 3) Call t_suspend() 4) Do t_continue() when async result is available 5) De-allocate pkg memory reserved for hashid and label 6) Do a t_relay() which forwards the sip message to another sip node. In the step (6) above, we see t_newtran() allocates one more time shared memory for the same transaction. We tried t_release() after step (4) to release the transaction as t_relay() anyways allocates new shared memory. Nothing helped. Please let me know what are the logs you would require to debug the same. I am attaching syslog for this run. Regards, Shankar

1 0

Regd. t_suspend() and t_continue()
by Shankar 13 Jan '14

13 Jan '14

Hi, We are trying out the t_suspend() and t_continue() in our test setup. We are facing memory leak ( both shm and pkg as per top command results). Please find below the scenario, 1) Do a t_newtran() 2) Allocate pkg memory for hashid and label. 3) Call t_suspend() 4) Do t_continue() when async result is available 5) De-allocate pkg memory reserved for hashid and label 6) Do a t_relay() which forwards the sip message to another sip node. In the step (6) above, we see t_newtran() allocates one more time shared memory for the same transaction. We tried t_release() after step (4) to release the transaction as t_relay() anyways allocates new shared memory. Nothing helped. Please let me know what are the logs you would require to debug the same. I am attaching syslog for this run. Regards, Shankar

1 0

Critical dialog - Bogus event 5 Problem - PRACK after 200OK
by Arik Lustig 13 Jan '14

13 Jan '14

Hello There, We are detecting errors quite often on our kamailio servers log: /usr/local/sbin/kamailio[17527]: CRITICAL: dialog [dlg_hash.c:782]: bogus event 5 in state 4 for dlg 0 x7fe0269a8be0 [1004:10921] with clid '034e8c8f-07daa8c0-001b56a3(a)192.168.210.2' and tags '07daa8c0-33aeaec-t-1' '9304808465080725853 ' After analyzing the issue We found that the root cause for those errors is when kamailio gets PRACK message after 200 OK (CSeq INVITE) in the same CALLID. To mention that those calls are successfuly established. This is due a complecated voice network. As for today we are getting around 100 errors per kamailio server every day. The number will increase to around 1000 errors per day in the near future. Two questions please: 1. Can Kamailio safety deal with those errors with no service effect 2. Is there a way to decrease the level of those errors from CRITICAL to INFO? As for today we are working in debug level 0 so no INFO written is written to the log in general. I will appriciate for any assistant on this topic please. Thanks, Arik.

2 2

The RPC Function in The TM Module
by 无愿无悔 13 Jan '14

13 Jan '14

Hello everyone: I want to know what is the parameter for the RPC function "tm.cancel" in the TM module ? I input the parameters such as : tm.cancel d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ 2 but it can accept the only parameter,and inform me "error 400-Callid and CSeq expected as parameters " . The context of the Callid.s is "d90be24a4553d076@Yml5cC5oYWlnzs5uZXQ". But the context of the CSeq.s is "if you get this string, you don't check rpc_scan return code!!!(very bad)". Could you give me an example ?

2 1

JSCommunicator feedback?
by Daniel Pocock 10 Jan '14

10 Jan '14

Hi, I'm just wondering if anybody has used JSCommunicator with Kamailio yet and if anybody has any feedback? http://jscommunicator.org/ If anybody publishes a dedicated blog about using them together I'll be happy to link to it I'm also looking for any feedback from anybody who may be able to take the CSS a little bit further. On the one hand, it would be nice to make it look a little bit more slick but on the other hand, the intention is for other people to drop it into their own sites where they have existing styles as well, so I'd prefer not to have it themed so aggressively that users have more work to do removing things. Regards, Daniel

2 4

Older Sip Phone ia hangup after 60sec with Realtime asterisk
by Noriyuki Hayashi 10 Jan '14

10 Jan '14

Hello, I am beginner using kamailio with much appreciated. Only one sip-phone is hang up after 60 seconds problem. This sip phone has no nat function at all.(SANYO SIP-2100) Grand Stream is works fine with kamailio. I would like give me your great advice with much appreciated. Environment. CentOS5.10, Asterisk-11.6.0 with PostgreSQL-9.2.5 as Realtime. Kamailio-4.1.0 Only Asterisk and PostgreSQL with older sip phone works fine. If Kamailio is running that registered is OK, But meetme(example) is hangup after 60 sec. I do not know "reINVITE or RTP" problem. Kamailio.cfg #!KAMAILIO #!enable postgresql #!define WITH_PGSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_DISPATCHER #!define WITH_ANTIFLOOD #!define WITH_MULTIDOMAIN #!define WITH_WITHINDLG #!define WITH_DEBUG #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_PGSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "postgres://postgres:password@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "postgres://asterisk:password@localhost/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 #log_facility=LOG_LOCAL0 log_facility=LOG_LOCAL7 fork=yes children=4 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #listen=udp:192.168.192.92 port=5060 mhomed=1 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk1.bindip = "192.168.192.92" desc "Asterisk IP Address" asterisk1.bindport = "5080" desc "Asterisk Port" asterisk2.bindip = "192.168.192.93" desc "Asterisk IP Address" asterisk2.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "192.168.192.92" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif #!ifdef WITH_DISPATCHER loadmodule "dispatcher.so" loadmodule "sqlops.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "name") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!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 #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #!ifdef WITH_DISPATCHER # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("sqlops","sqlcon", "ca=>postgres://asterisk:password@localhost/kamailio") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # 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"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sip_devices", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if(ds_is_from_list()) { return 1; } else { return -1; } } # Send to Asterisk route[TOASTERISK] { if($(au{s.len})<=5) { $var(setid) = 0; xlog("SCRIPT: Connected Asterisk #0 - using set $var(setid) \n"); } else { $var(setid) = 9; xlog("SCRIPT: Connected Asterisk #9 - using set $var(setid) \n"); } # failover dispatching on set determined above if(!ds_select_dst($var(setid), "8")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($(au{s.len})<=5) { $var(rip) = $sel(cfg_get.asterisk1.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk1.bindport); } else { $var(rip) = $sel(cfg_get.asterisk2.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk2.bindport); } $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } #!endif *** Test call to meetme Logs. **** sip1*CLI> sip set debug on sip1*CLI> SIP Debugging re-enabled sip1*CLI> sip set debug on sip1*CLI> Name/username Host Dyn Forcerport ACL Port Status Description Realtime 99206/99206 192.168.192.92 D N 5060 OK (515 ms) Cached RT 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] sip1*CLI> -- Executing [901@99:1] Answer("SIP/99206-00000000", "") Audio is at 15506 sip1*CLI> Adding codec 100003 (ulaw) to SDP sip1*CLI> Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP sip1*CLI> <--- Reliably Transmitting (NAT) to 192.168.192.92:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9 Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes> From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 INVITE Session-Expires: 120;refresher=uas Contact: <sip:901@192.168.192.92:5080> Content-Type: application/sdp Require: timer Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> sip1*CLI> Retransmitting #1 (NAT) to 192.168.192.92:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9 Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes> From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 INVITE Session-Expires: 120;refresher=uas Contact: <sip:901@192.168.192.92:5080> Content-Type: application/sdp Require: timer Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> ACK sip:901@192.168.192.92:5080 SIP/2.0 From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e Max-Forwards: 16 Contact: <sip:99206@192.168.192.190:5060> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901@192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5 Content-Length:0 <-------------> --- (11 headers 0 lines) --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> ACK sip:901@192.168.192.92:5080 SIP/2.0 From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e Max-Forwards: 16 Contact: <sip:99206@192.168.192.190:5060> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901@192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5 Content-Length:0 <-------------> --- (11 headers 0 lines) --- sip1*CLI> -- Executing [901@99:2] Wait("SIP/99206-00000000", "1") sip1*CLI> > 0x17aa0bd0 -- Probation passed - setting RTP source address to 192.168.192.190:17096 sip1*CLI> -- Executing [901@99:3] Authenticate("SIP/99206-00000000", "5963") sip1*CLI> -- <SIP/99206-00000000> Playing 'agent-pass.gsm' (language 'ja') sip1*CLI> -- <SIP/99206-00000000> Playing 'auth-thankyou.gsm' (language 'ja') sip1*CLI> -- Executing [901@99:4] MeetMe("SIP/99206-00000000", "99901,pM") == Parsing '/etc/asterisk/meetme.conf': Found sip1*CLI> -- Created MeetMe conference 1023 for conference '99901' sip1*CLI> -- <SIP/99206-00000000> Playing 'conf-onlyperson.gsm' (language 'ja') sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000 sip1*CLI> -- Stopped music on hold on SIP/99206-00000000 sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000 sip1*CLI> Audio is at 15506 Adding codec 100003 (ulaw) to SDP Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.192.92:5060: INVITE sip:99206@192.168.192.190:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport Max-Forwards: 70 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Contact: <sip:901@192.168.192.92:5080> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.6.0 Session-Expires: 120;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, BYE X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport=5080 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 102 INVITE Server: kamailio (4.1.0 (x86_64/linux)) Content-Length: 0 --- > [INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid") VALUES ('2014-01-06 15:49:12','"Richard Nough" <99206>','99206','901','99','SIP/99206-00000000','MeetMe','99901,pM',60,60,'ANSWERED',3,'nori@wats','1388990952.0')] <--- SIP read from UDP:192.168.192.92:5060 ---> SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK339a9eb8;rport=5080 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 103 BYE Server: kamailio (4.1.0 (x86_64/linux)) Content-Length: 0 I hope you have a great 2014. Kind regards, Nori

2 4

Max-Forwards severly decreased when invite is relayed
by Grant Bagdasarian 10 Jan '14

10 Jan '14

Hello, Using Kamailio 4.1. I'm having some problems with the Max-Forwards header being decremented by almost 50 when Kamailio relays the INVITE to the next hop. The INVITE is received by Kamailio with the value of 65, but when sent to the next hop it has the value of 16. I'm not altering the max-forwards header value in any way in the config script. At least not explicitly. Are there any known constructs which indirectly affect the Max-Forwards header? 10.0.0.16 is the Kamailio server. U 10.0.0.10:5060 -> 10.0.0.16:5060 INVITE sip:3110123456@10.0.0.16:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.10;rport;branch=z9hG4bKeg5g6U4Dc5HeS. Max-Forwards: 65. From: "31765123456" <sip:31765123456@10.0.0.10>;tag=9Z7avj87Bc2gm. To: <sip:3110123456@10.0.0.16:5060>. Call-ID: e089c971-f3a8-1231-c2bb-000c2941b070. CSeq: 54291565 INVITE. Contact: <sip:mod_sofia@10.0.0.10:5060>. User-Agent: CM Carrier SBC. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 205. U 10.0.0.16:5060 -> 10.0.0.31:5060 INVITE sip:3110123456@10.0.0.25:5060 SIP/2.0. Record-Route: <sip:10.0.0.16;lr=on>. Via: SIP/2.0/UDP 10.0.0.16;branch=z9hG4bKe501.5459c149e5fa2c6c723149661017aa3d.0. Via: SIP/2.0/UDP 10.0.0.10;rport=5060;branch=z9hG4bKeg5g6U4Dc5HeS. Max-Forwards: 16. From: "31765123456" <sip:31765123456@10.0.0.10>;tag=9Z7avj87Bc2gm. To: <sip:3110123456@10.0.0.16:5060>. Call-ID: e089c971-f3a8-1231-c2bb-000c2941b070. CSeq: 54291565 INVITE. Contact: <sip:mod_sofia@10.0.0.10:5060>. User-Agent: CM Carrier SBC. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 205.

2 2

attrs field in "kamctl fifo ds_list"
by Benjamin Henrion 09 Jan '14

09 Jan '14

Hi, Any idea what attrs are in dispatcher.cfg? When I do a kamctl fifo ds_list, I can put some strings in this field. Is it used for labelling of the SIP URIs? The documentation is a bit obscure for me. If I put hello as the last item on dispatcher.cfg, I see it when I do a ds_list: $ kamctl fifo ds_list SET:: 99 URI:: sip:127.0.0.1:5060 flags=AP priority=0 attrs=hello Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-4148403 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

2 1

Kamailio v4.1.1 Released
by Daniel-Constantin Mierla 09 Jan '14

09 Jan '14

Hello, Kamailio SIP Server v4.1.1 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.1. For more details about version 4.1.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/01/kamailio-v4-1-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Releasing v4.1.1
by Daniel-Constantin Mierla 09 Jan '14

09 Jan '14

Hello, I am considering to release v4.1.1 out of branch 4.1 this week on Thursday -- there were couple of fixes since we did v4.1.0. If there are any open issues that should get addresses, reply here. Also, for developers, if there are patches only on master branch that should be pushed on 4.1 as well, do it as soon as possible. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

Kamdbctl create issue
by Owais ul Haq 09 Jan '14

09 Jan '14

Hello all, I am installing Kamailio-4.1.0 on my system. I get this error while creating 'kamdbctl' file. [root@ss3 sbin]# ./kamdbctl create ERROR: DATABASE ENGINE NOT SPECIFIED, PLEASE SETUP ONE IN THE CONFIG SCRIPT. Can some one please help in solving this error. Any help will be appreciated. Cheers, Kamilio Newbie.

2 1

Kamailio-4.1.0 Installation Issues
by Owais ul Haq 09 Jan '14

09 Jan '14

Hello all, I am installing Kamailio-4.1.0 on my system. I get this error while creating 'kamdbctl' file. Can some one please help in solving this error. Any help will be appreciated. Cheers, Kamilio Newbie.

2 1

question about nat_uac_test
by Brian Davis 08 Jan '14

08 Jan '14

When using nat_uac_test("3"), there is a particular REGISTER message that my kamailio server receives where the source and via addresses are different, but this function seems to return false (and then no nat correction is applied). I have included the register message below. Is there an obvious reason this is not triggering the nat_uac_test? Or can anyone share how they would usually debug a situation like this? Any help/advice would be appreciated. Thank you! Example captured Register message: REGISTER sip:test1.test.com:5060 SIP/2.0 Call-ID: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx CSeq: 1418 REGISTER From: "Joe" <sip:xxxyyyzzzz@test1.test.com>;tag=2163936191 To: "Joe" <sip:xxxyyyzzzz@test1.test.com> Via: SIP/2.0/UDP 96.xxx.xxx.xxx:33745;rport;branch=z9hG4bKf5s1p`n3TRv5TZx5RXy.RVv+JPz8Nat*UX!8KRx4SRx Via: SIP/2.0/UDP 192.xx.xxx.xxx:33745;branch=z9hG4bKeb263246c44095f072d8167dd0c7987a343134;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "Joe" <sip:xxxyyyzzzz@192.xx.xxx.xxx:33745;transport=udp> Expires: 3600 Authorization: Digest username="xxxyyyzzzz",realm="test1.test.com ",nonce="UsDqxVLA6ZlvIrO3TsqCtm9X3vqY8405",uri="sip:test1.test.com:5060 ",response="769d1e6f2fd36cf040503f9f079279a0" Content-Length: 0 Associated lines from kamailio log: [kamailio]# grep 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxxkamailio.log Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|start|recieved UDP request REGISTER sip:test1.test.com Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|source 96.xxx.xxx.xxx:33745 Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|from sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|to sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|originated from external sources Dec 30 03:33:45 sip-01 kamailio[20489]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|end|issued new auth challenge to new registration attempt Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|start|recieved UDP request REGISTER sip:test1.test.com:5060 Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|source 96.xxx.xxx.xxx:33745 Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|from sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|to sip:xxxyyyzzzz@test1.test.com Dec 30 03:33:45 sip-01 kamailio[20495]: INFO: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|originated from external sources Dec 30 03:33:45 sip-01 kamailio[20495]: DEBUG: <script>: 3b0400ca43e28f78f3e6dc945a084b88(a)192.xx.xxx.xxx|log|authenticated xxxyyyzzzz(a)test1.test.com via cached SIP creds

2 3

Kamailio 4.1.x Installation
by Owais ul Haq 08 Jan '14

08 Jan '14

Hello all, I am currently using Kamailio-3.1.1 installed on a fedora machine. I want to install updated version Kamailio-4.1.x. Can someone give me some link where I can find about installation steps and pre-requisites etc. Thanks for helping. Regards, Kamailio Newbie :)

3 2

kamailio crash after sqlops failures
by Uri Shacked 08 Jan '14

08 Jan '14

Hi kamailio version 3.2.0 16M configured has private memory and 4G as shared memory. using sqlops to query the DB. the DB was down and kamailio crashed after some time with the following log under 1. /var/log/messages "kernel: kamailio[PID]: segfault at 5 ip 00007fdacc65e175 sp 00007fff85897810 error 4 in tm.so[7fdacc630000+79000]" 2./var/log/kamailio " /usr/local/sbin/kamailio[12033]: ERROR: <core> [db_query.c:156]: error while submitting query /usr/local/sbin/kamailio[12033]: ERROR: sqlops [sql_api.c:580]: cannot do the query /usr/local/sbin/kamailio[12055]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 14 /usr/local/sbin/kamailio[11966]: ALERT: <core> [main.c:785]: child process 12034 exited by a signal 11 /usr/local/sbin/kamailio[11966]: ALERT: <core> [main.c:788]: core was generated" (core was not generated due to lack of permissions on linux..... but now it is ok) So, anyone has an idea where to start? BR, Uri

2 4

gateway authentication with the lcr module
by Rene Montilva 08 Jan '14

08 Jan '14

Hi List is there any option with lcr module to authenticate gateways with digest method for outbound calls?, or whats the best practice for that scenario. i have a sip user from my sip provider, it register with uac module for incoming calls, but i want outbound calls too Best Regards

3 2

Re: [SR-Users] Kamailio 4.1, Debian Squeeze, RTPProxy 1.2.1 Segfault [SOLVED]
by Jr Richardson 08 Jan '14

08 Jan '14

On Mon, Dec 30, 2013 at 6:08 PM, Jr Richardson <jmr.richardson(a)gmail.com> wrote: >> >> Hello, >> >> you should enable core dumping - that should be via running 'ulimit -c >> unlimited' before you start rtpproxy. >> >> If you get the coredump, then grab the backtrace with gdb, it should >> show where rtpproxy crashed. With that info, maybe we can spot a fix for it. >> >> Hopefully rtpproxy package is with debug symbols, if not, the look for a >> rtpproxy-dbg package and install it if found. >> >> Cheers, >> Daniel >> >> On 20/12/13 18:59, Jr Richardson wrote: >>> Hi All, >>> root@sip-router3-ve206:/etc/kamailio# kamailio -V >>> version: kamailio 4.1.0 (x86_64/linux) 350d2e >>> Doing some testing, can't seem to get rtpproxy to not segfault. I've >>> loaded version from deb-squeeze pkg, from source >>> http://b2bua.org/chrome/site/rtpproxy-1.2.1.tar.gz and from >>> git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy >>> <http://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy> (which >>> should be the latest) but all versions segfault as soon as a call sets up. >>> I've configured each versions control socket for both udp or unix, >>> kamailio starts and sees the rtpproxy fine with no errors, but when a >>> call hits, rtpproxy segfaults. >>> I've run rtpproxy in the forground with degug and get this response >>> (IP's washed x.x.x): >>> ------------------ >>> DBUG:handle_command: received command "7505_12 USIEc0,101 >>> 0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76 x.x.x.76 23388 as33dd7c98;1" >>> INFO:handle_command: new session >>> 0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76, tag as33dd7c98;1 requested, >>> type strong >>> Segmentation fault >>> ------------------ >>> root@sip-router3-ve206:/etc/kamailio# more /etc/default/rtpproxy >>> # Defaults for rtpproxy >>> # The control socket. >>> CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" >>> # To listen on an UDP socket, uncomment this line: >>> #CONTROL_SOCK="udp:127.0.0.1:7722 <http://127.0.0.1:7722>" >>> # Additional options that are passed to the daemon. >>> EXTRA_OPTS="-l x.x.x.20" >>> kamailio.cfg: >>> loadmodule "rtpproxy.so" >>> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722 >>> <http://127.0.0.1:7722>") >>> inside route[] >>> >>> add_path_received(); >>> rtpproxy_manage("cwei"); >>> record_route(); >>> Any guidance on further identifying the issue? >>> Thanks. >>> JR > > I'm not very familiar with gdb, but did manage to get a core dump and > here are the results: > ------------------------------------------- > root@rtpproxy:/usr/src/rtpproxy-git# gdb > /usr/src/rtpproxy-git/rtpproxy -c /usr/src/rtpproxy-git/core > GNU gdb (GDB) 7.0.1-debian > Copyright (C) 2009 Free Software Foundation, Inc. > License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> > This is free software: you are free to change and redistribute it. > There is NO WARRANTY, to the extent permitted by law. Type "show copying" > and "show warranty" for details. > This GDB was configured as "x86_64-linux-gnu". > For bug reporting instructions, please see: > <http://www.gnu.org/software/gdb/bugs/>... > Reading symbols from /usr/src/rtpproxy-git/rtpproxy...done. > [New Thread 11256] > warning: Can't read pathname for load map: Input/output error. > Reading symbols from /lib/libpthread.so.0...(no debugging symbols found)...done. > Loaded symbols for /lib/libpthread.so.0 > Reading symbols from /lib/libm.so.6...(no debugging symbols found)...done. > Loaded symbols for /lib/libm.so.6 > Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. > Loaded symbols for /lib/libc.so.6 > Reading symbols from /lib64/ld-linux-x86-64.so.2...(no debugging > symbols found)...done. > Loaded symbols for /lib64/ld-linux-x86-64.so.2 > Reading symbols from /lib/libnss_compat.so.2...(no debugging symbols > found)...done. > Loaded symbols for /lib/libnss_compat.so.2 > Reading symbols from /lib/libnsl.so.1...(no debugging symbols found)...done. > Loaded symbols for /lib/libnsl.so.1 > Reading symbols from /lib/libnss_nis.so.2...(no debugging symbols found)...done. > Loaded symbols for /lib/libnss_nis.so.2 > Reading symbols from /lib/libnss_files.so.2...(no debugging symbols > found)...done. > Loaded symbols for /lib/libnss_files.so.2 > Core was generated by `./rtpproxy -f -l x.x.x.x -s udp:x.x.x.x 7722 -p > /var/run/rtpproxy/rtp'. > Program terminated with signal 11, Segmentation fault. > #0 create_twinlistener (cf=0x7fff957ec2d0, ia=0x0, port=<value > optimized out>, fds=<value optimized out>) > at rtpp_command.c:97 > 97 fds[i] = socket(ia->sa_family, SOCK_DGRAM, 0); > (gdb) bt > #0 create_twinlistener (cf=0x7fff957ec2d0, ia=0x0, port=<value > optimized out>, fds=<value optimized out>) > at rtpp_command.c:97 > #1 create_listener (cf=0x7fff957ec2d0, ia=0x0, port=<value optimized > out>, fds=<value optimized out>) > at rtpp_command.c:143 > #2 0x000000000040967d in handle_command (cf=0x7fff957ec2d0, > controlfd=<value optimized out>, > cmd=0x7f73bf148d00, dtime=<value optimized out>) at rtpp_command.c:837 > #3 0x000000000040bc9d in process_commands (cf=0x7fff957ec2d0, > controlfd_in=3, dtime=<value optimized out>) > at rtpp_command_async.c:63 > #4 0x000000000040bd75 in rtpp_cmd_queue_run (arg=<value optimized > out>) at rtpp_command_async.c:92 > #5 0x00007f73bff6e8ca in start_thread () from /lib/libpthread.so.0 > #6 0x00007f73bfa53b6d in clone () from /lib/libc.so.6 > #7 0x0000000000000000 in ?? () > (gdb) > --------------------------------------------------------------------------------- > I could not find an rtpproxy debug package and the install notes do > not tell how to compile with debugging symbols. > > Thanks. > > JR So apparently rtpproxy doesn't like to be told to use 2 interfaces in the flags (ie or ei) when only 1 interface is actually enabled for use. This was a case of RTFM! RTPProxy seems to be working fine, once I got the correct config parameters in kamailio. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Cassandra module
by Carlos Ruiz Díaz 08 Jan '14

08 Jan '14

Hi, does anybody know if it is possible to use Kamailio's built-in maintenance tools with the db schema as tables in Cassandra? I'm trying to create the "subscriber" table in a Cassandra cluster and maintain it using kamctl options for new users and passwords. I tried modifying the DBENGINE in kamctlrc to DB_CASSANDRA, CASSANDRA, DBCASSANDRA, without luck. Regards, -- Carlos http://caruizdiaz.com +595981146623

2 1

tag column on address table
by Kelvin Chua 07 Jan '14

07 Jan '14

is the peer_tag_avp used on allow_source_address_group? Kelvin Chua

3 11

SIP Security Architectural Question to Use RTP/Media Proxy or Not?
by Jr Richardson 07 Jan '14

07 Jan '14

Hi All, Background: We are a service provider offering VoIP/Data services to business customers. All hosted VoIP systems and Customers are mostly on-net, VoIP systems not exposed to the Internet, but all hosted PBX's do have public IP address. I do have some Customers with off-net phones/users so I basically white list their IP's so the phones can register back to their hosted PBX. This works well and keeps SIP attack vectors to a minimum. I've been working on a single point of registration Kamailio server to backend PBX's so I can further control public Internet access to hosted PBX's. I've got this working in the lab but have some concerns about RTP streams. I know I can use a RTP/Media Proxy to also have a single point of entry for media streams to the the backend PBX's but don't believe this to be the best method. Researching SBC's and what I know about SIP and RTP Streams, it's best to have media controlled via the B2BUA (Asterisk in this case) and since all my hosted PBX's have public IP's there would be no compelling reason to proxy RTP adding another hop, latency and point of failure other than for security. I'm not transcoding media or doing anything outside of the capability of the B2BUA as far as media goes. Question: Would it be prudent to open UDP media ports from Internet to PBX's on a case-by-case basis, basically white listing media streams or is there any attack vulnerability with UDP in the media port range or should I open up media port range to all PBX's and not worry about attacks. Are there any UDP Media exploits that I should be concerned with, or UDP flood attacks that could DOS my hosted PBX's? Thanks for any feedback. JR -- JR Richardson Engineering for the Masses

4 3

Got SIP response 400 "Bad From URI"
by arun Jayaprakash 06 Jan '14

06 Jan '14

Hello, I have set up two extensions in my network, both have identical configurations. When I call from ext A to ext B I have no problem; ext B rings and I can hear audio. Now when I call from ext B to ext A I get the following error: "Got SIP response 400 "Bad From URI" back from 172.10.30.15". Could someone let me know why this happens, what could be the issue and what I need to check for? Thank you for your help. Arun Jayaprakash

2 1

Kamailio World 2014 - Call for Speakers
by Daniel-Constantin Mierla 06 Jan '14

06 Jan '14

Hello, I would like to announce that call for speakers at Kamailio World 2014 is now open. Those interested in presenting at the event have to submit the proposal via the web form at: - http://conference.kamailio.com/k02/call-for-speakers/ Have a great 2014, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Got SIP response 400 "Bad From URI"
by arun Jayaprakash 05 Jan '14

05 Jan '14

Hello, I have set up two extensions in my network, both have identical configurations. When I call from ext A to ext B I have no problem; ext B rings and I can hear audio. Now when I call from ext B to ext A I get the following error: "Got SIP response 400 "Bad From URI" back from 172.10.30.15". Could someone let me know why this happens, what could be the issue and what I need to check for? Thank you for your help. Arun Jayaprakash

1 0

t_reply in failure route with dialog module
by Efelin Novak 03 Jan '14

03 Jan '14

Hi, when I use t_reply("505", "Error"); in my failure route, the response is not forwarded and following is written into a log: kamailio[26216]: WARNING: tm [t_lookup.c:1559]: t_unref(): WARNING: script writer didn't release transaction plus next line is written exactly 416000 times into a log afterwards: kamailio[32685]: CRITICAL: dialog [dlg_hash.c:794]: log_next_state_dlg(): bogus event 4 in state 5 for dlg 0xb4af6588 [2575:7017] with clid '121d44f0-6555f4c8' and tags 'd12546d053aadc68o2' '' My point is to change the incoming code from users and append a Q.850 reason code. Is there any other way how to do this or a way how to fix this? I'm using Kamilio 4.0.4 on Debian 7.1 The code is as follows: failure_route[MANAGE_FAILURE] { if (t_is_canceled()) { exit; } if($T_reply_code == 408 && isflagset(10)) { xlog("Ringing timeout"); append_to_reply("Reason: Q.850;cause=28\r\n"); t_reply("505", "Error"); } }

3 10

Missing Route headers in locally generated BYE
by Efelin Novak 03 Jan '14

03 Jan '14

Hi, I locally generate BYE using dlg_end_dlg. When I want to end a call that is "transport layer" bridged, the BYE is not sent to first hop in route_set but directly to the endpoint. In such BYE there are no Route headers. In non-bridging calls Routes are correctly placed and the message is routed to the first "hop". When the error happens, this is written to a log: WARNING: rr [loose.c:821]: after_loose(): no socket found for match second RR Here ([SR-Users] no socket found for match second RR) I have read this is only a warning, but in my configuration it seriously influences the message routing. My setup is phone1(192.168.10.3) <--TCP--> kamailio1(192.168.10.2) <--UDP--> kamailio2(192.168.5.3) <--UDP--> phone2 On kamailio1 I generate dlg_end_dlg and the BYE is sent to phone1 and phone2 directly. I'm using Kamailio 4.0.4 on Debian machines. How can I make the Kamailio1 to send the BYE to kamailio2 in the transport layer bridging scenario? Do I have some misconfiguration or this is not a correct behaviour? Thanks for answer Efelin

4 7

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users