- sr-users - kamailio.org

Problem adding json module
by Joao Alves 07 Aug '15

07 Aug '15

Hi, Have just installed the kamailio 4.3 (from github) and was trying to include the additional modules that I require. I've edited the modules.lst and changed the extra modules to compile "include_modules= db_mysql utils json". During compilation (using "make all") getting the following warnings in relation with the json module. CC (gcc) [M json.so] json_funcs.o json_funcs.c: In function 'json_get_field': json_funcs.c:58:2: warning: implicit declaration of function 'is_error' [-Wimplicit-function-declaration] json_funcs.c:64:2: warning: 'json_object_object_get' is deprecated (declared at /usr/local/include/json-c/json_object.h:303) [-Wdeprecated-declarations] And later, when running the config file: ERROR: <core> [sr_module.c:574]: load_module(): could not open module </usr/local/lib64/kamailio/modules/json.so>: libjson-c.so.3: cannot open shared object file: No such file or directory 0(2344) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 195, column 12-20: failed to load module ERROR: bad config file (1 errors) I have already installed json-c modules and confirmed this installation by doing: root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --cflags json-c -I/usr/local/include/json-c root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --libs json-c -L/usr/local/lib -ljson-c root@ID14337:/usr/local/include/json-c# ls arraylist.h json.h json_inttypes.h json_object_private.h linkhash.h bits.h json_c_version.h json_object.h json_tokener.h printbuf.h debug.h json_config.h json_object_iterator.h json_util.h random_seed.h root@ID14337:/usr/local/include/json-c# cd ../../lib root@ID14337:/usr/local/lib# ls libjson-c.a libjson-c.la libjson-c.so libjson-c.so.3 libjson-c.so.3.0.0 pkgconfig python2.6 python2.7 root@ID14337:/usr/local/lib# So any ideas of what could be causing this to fail? Thanks Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

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Kamailio w/ external routing server
by Eric Babcock 07 Aug '15

07 Aug '15

Hey Everyone, I've been banging my head against the desk for days trying to get this working, and I'm sure it's something simple that I'm not doing, but maybe someone else can add some input. Here is the best way to describe my call flow. Customer -> Kamailio -> Routing Server (302) -> Kamailio -> Vendors (from the 302 header) I have tried to configure dispatch, but that hasn't worked out for me. My routing server does the authentication, LRN lookup, and provides the vendor routes in the sip header. So when I pass the call to the routing server, it has to contain the customer information. This setup has to be stateful so I can write CDRs for billing to MySQL. The reason I tried using dispatch, is because I have multiple routing servers I would like to load balance across or even failover if one goes down. Can anyone help me out, or at least help point me in the right direction? Thanks! -Eric -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-w-external-routing-server-… Sent from the Users mailing list archive at Nabble.com.

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Errors when starting Kamailio with Websocket Module
by Babu Raja 07 Aug '15

07 Aug '15

Hello, I incuded the module for websocket into my Kamailio 4.3 on my Ubuntu 12.04 server environement. For installing the packages and websocket I followed the instructions from the following link: http://www.kamailio.org/wiki/packages/debs http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket After starting the Kamailio I got 26 errors like: can not find command "handle_ruri_alias" or Syntax errors for: route[REGISTRAR] => square brackets.. ($du == "") => parameter "==" unknown.. the whole error log I will paste later.. Can anyone help me?

3 4

RLS Notify sent to wrong transport
by M S 07 Aug '15

07 Aug '15

Hi, I have a Kamailio v4.3.0 with SVN Rev. c6aa95 running an RLS based presence setup. Everything works fine when end-user has UDP transport, however, if user has TCP or TLS transport then i aggregated NOTIFY sent by RLS gives send error, -- WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:X.X.X.X:5061, to udp:Y.Y.Y.Y:12345) -- (X.X.X.X is server IP and Y.Y.Y.Y is end user IP). Not sure if it is configuration issue or some bug, so posting it to both mailing list. Please help. Thank you.

2 8

Kamailio 4.3.1 and Debian 7
by Mathieu Boyer 07 Aug '15

07 Aug '15

Hello, i've got big issues since upgrade of Debian packages (debian repository http://deb.kamailio.org/kamailio wheezy)<http://deb.kamailio.org/kamailio> on 4.3.1. Started with rr.so can't load without outbound.so, location.so find version mismatch (6 vs 8) etc ... Have you got a quick workaround in order to make it run properly ? dpkg -l |grep kamailio ii kamailio 4.3.1+wheezy amd64 very fast and configurable SIP proxy ii kamailio-extra-modules:amd64 4.3.1+wheezy amd64 extra modules for Kamailio ii kamailio-mysql-modules:amd64 4.3.1+wheezy amd64 MySQL database connectivity module for Kamailio ii kamailio-outbound-modules:amd64 4.3.1+wheezy amd64 Outbound module for Kamailio ii kamailio-utils-modules:amd64 4.3.1+wheezy amd64 Provides a set utility functions for Kamailio Error logs : Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4083]: ERROR: dispatcher [dispatcher.c:768]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'server_id' in 'field list' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [udomain.c:424]: preload_udomain(): db_query (1) failed Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [ul_mod.c:439]: child_init(): child(1): failed to preload domain 'location' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [sr_module.c:900]: init_mod_child(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [pt.c:329]: fork_process(): init_child failed for process 1, pid 4085, "udp receiver child=0 sock=XXXX:5063"? Thank you for your help BR Mathieu

3 2

Rtpengine/rtpproxy and zrtp
by Alexandru Covalschi 07 Aug '15

07 Aug '15

Sorry if writing to wrong mailing list, I am very limited to traffic now amd don't know if there is any for rtpproxy/rtpengine. My question is - can they support ZRTP at least in pass-through mode? Will rtpengine fail on trying to recognize unknown SDP fields?

2 2

Kamailio v4.2.6 Released
by Daniel-Constantin Mierla 06 Aug '15

06 Aug '15

Hello, Kamailio SIP Server v4.2.6 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.6 (or to 4.3.x series). For more details about version 4.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.1. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

2 3

Kamailio Redis support
by Solutions Solutions 06 Aug '15

06 Aug '15

Hi All, i want to replace MySQL with redis database in kamalio,but in kamalio supports transactions with MySQL ,ie saving data in MySql DB.but while trying to run kamailio with redis ,transaction are not happening,is there any module that i have to enable in kamalio for No SQL databases transaction support.

2 1

Re: [SR-Users] Remote SIP Registration
by Keith 05 Aug '15

05 Aug '15

Hi, I have a carrier who require registration, I have looked at the uac module but can't see how to do this. Anyone point me in the right direction? Thanks Keith

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Difference between corex module's 'alias_subdomains' and 'alias' core parameter
by Iwan Aucamp 05 Aug '15

05 Aug '15

>From the descriptions these seem similar: corex alias_subdomains ( http://kamailio.org/docs/modules/devel/modules/corex.html#idp18017160): Register a domain and all its sub-domains to match the “myself” condition. It can be set many times. Its full format is: 'proto:domain:port', allowing to set restrictions on protocol and port as well. Protocol and port are optional. alias core parameter ( http://www.kamailio.org/wiki/cookbooks/devel/core#alias): Parameter to set alias hostnames for the server. It can be set many times, each value being added in a list to match the hostname when 'myself' is checked. Is the only difference that corex alias_subdomains can include 'proto' ? -- Iwan Aucamp

2 1

Question about how location server_id is supposed to be used.
by Ding Ma 05 Aug '15

05 Aug '15

Looking for some doc that describes how the new columns in 4.3.x location table is supposed to be used. Can find some email discussions about this, but seems no formal doc exists. Would appreciate if someone can provide an explanation. Thanks

3 3

Re: [SR-Users] R: Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 05 Aug '15

05 Aug '15

Indeed usually there are few options for same functionality, good to hear you found a solution. Cheers, Daniel On 03/08/15 14:53, Marino Mileti wrote: > Thanks Daniel, > > I've found another way using a seturi for mediaserver that will be executed > after all branch (using t_load_contacts and t_next_contacts) > In this way I can complete my scenario in the right way > > -----Messaggio originale----- > Da: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Per conto di > Daniel-Constantin Mierla > Inviato: lunedì 3 agosto 2015 12:37 > A: Kamailio (SER) - Users Mailing List > Oggetto: Re: [SR-Users] Mutiple branch forking and voicemail > > Hello, > > On 31/07/15 11:31, Marino Mileti wrote: >> Hello, >> >> I would like to implement a voicemail services but i'm trying >> everything without good results :( >> >> My scenario is: >> >> - A make a parallel forking to B(1001) and C(1002) >> - If timeout, call is routed to voicemail >> >> I've done something like this...very simple >> >> seturi("sip:1001@172.20.40.103"); >> >> append_branch("1002(a)172.20.40.103"); >> route(RELAY) >> >> on failure_route.. >> if (t_check_status("486|408")) { >> $ru = "sip:8000@172.20.40.103:5090"; >> >> route(RELAY); >> >> exit; >> >> } >> >> The scenario is working but on log there are some error... >> t_should_relay_response >> >> I would like that when a branch reach the failure route, all other >> branches will be cancelled..and call is routed to VM. >> Can anyone help me? >> >> > When the failure_route is executed, no branch is active anymore. > > To be able to assist further, you have to send here the errors you get in > the log, to understand what is wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Maintaining order of messages in a sip dialog on forwarding
by Daniel Tryba 05 Aug '15

05 Aug '15

Since switching to 4.2.5 (from 3.x) a customer has problems with a buggy user agent (IMHO). What I am seeing now is: provider kamailio customer INVITE1-> INVITE1-> <-Trying <-Trying <-200 OK <-200 OK ACK1-> INVITE2-> INVITE2-> ACK1-> ACK1 arrives 0.007s before INVITE2, but INVITE2 is send 0.004s before ACK1. It looks like this wasn't happening with Kamailio 3.x, it worked fine then (apparently). Can this reordering be (gracefully) prevented somehow (ignoring the fact that during transport reordering can happen)? A Q&D hack looks to be just usleep the reinvite a little. The endpoint is essentially ignoring the reinvite before (and after) the ACK. After a timeout INVITE2 is canceled and after that INVITE1 will be ended with a BYE from provider. kamailio-customer communication is UDP provider-kamailio is tcp when message >= 1300b. INVITE1 is TCP, INVITE2 and ACK1 are UDP.

2 5

VIA header question.
by Chad 04 Aug '15

04 Aug '15

Hi list, I need a little help, I am a business owner trying to get Kamailio up and running as a SIP load balancer. I hired a Kamailio consultant to help me do so, but Kamailio is not working and I am getting conflicting information. My Kamailio consultant and my VOIP provider are telling me 2 different things and I don't know which one is right. Kamailio sends SIP traffic to the VOIP provider with 2 VIA headers like this (in this order): Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. The VOIP provider says that is incorrect because they are supposed to reply back to the topmost VIA header so they reply to the 10.10.10.254 IP (which is not public) and the call ends. The VOIP provider says Kamailio should send the VIA headers like this instead: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. My Kamailio consultant says the way we are sending it is right and that the VOIP provider is processing the call incorrectly. I read that the SIP proxy is supposed to remove the internal header from the 1st example above based on this RFC: https://tools.ietf.org/html/rfc3261#section-16.7 Item: 3. Via "The proxy removes the topmost Via header field value from the response." If that applies to this situation (which I don't know if it does) then Kamailio should be removing the 10.10.10.254 VIA line and only sending 1 VIA header like this: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Which would sort of make the VOIP provider right in that the topmost VIA line would then be the external IP, but how they said to fix it (reversing the VIA lines) is wrong. Does anyone know what the right answer is here? Please let me know. THANKS for your help. -- ^C

3 3

MSILO: Archiving every message
by SamyGo 03 Aug '15

03 Aug '15

Hi, Tried finding anything as weird as I am trying to do on mailing list but couldn't. The idea is to save a copy of the messages on database. MSILO module can keep the message as long as the destination user is offline, or doesn't support method:MESSAGE or expiry timeout. Any ideas will be appreciated. Best Regards, Sammy

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Re: [SR-Users] Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 03 Aug '15

03 Aug '15

Hello, On 31/07/15 11:31, Marino Mileti wrote: > Hello, > > I would like to implement a voicemail services but i'm trying everything > without good results :( > > My scenario is: > > - A make a parallel forking to B(1001) and C(1002) > - If timeout, call is routed to voicemail > > I've done something like this...very simple > > seturi("sip:1001@172.20.40.103"); > > append_branch("1002(a)172.20.40.103"); > route(RELAY) > > on failure_route.. > if (t_check_status("486|408")) { > $ru = "sip:8000@172.20.40.103:5090"; > > route(RELAY); > > exit; > > } > > The scenario is working but on log there are some error... > t_should_relay_response > > I would like that when a branch reach the failure route, all other branches > will be cancelled..and call is routed to VM. > Can anyone help me? > > When the failure_route is executed, no branch is active anymore. To be able to assist further, you have to send here the errors you get in the log, to understand what is wrong. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

2 1

www.kamailio.org unreachable via ipv6
by Daniel Tryba 03 Aug '15

03 Aug '15

(www.)kamailio.org is unreachable at this time (the whole morning) on 2a00:d60:0:400::2. ipv4 is responsing. Never noticed whether connectivity via ipv6 was working in the past or this is new and misconfigured.

2 1

Announcement: Kamailio is now systemd-rtc-server
by Alex Balashov 03 Aug '15

03 Aug '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina,was quick to summarise the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Check if user is registered before reply is routed
by Dmytro Bogovych 01 Aug '15

01 Aug '15

Greetings. Sorry for previous false letter. I want to check if user is registered before reply is routed to its target user agent. Imho it should be like this: onreply_route[MANAGE_REPLY] { if ($rm=="INVITE") { if (registered("location", $ru)) { # Do job - pause for 15 seconds and give chance to reregister } } } But kamailio complains on registered function 0(6274) ERROR: <core> [cfg.y:3295]: yyparse(): misused command registered 0(6274) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1121, column 35: Command cannot be used in the block Documentation confirms - registered() function can be used in REQUEST_ROUTE and FAILURE_ROUTE only. Is it possible to check if user is registered when reply is processed in scripts? Thank you!

4 3

Interpretation of Contact header
by Joachim Büchse 30 Jul '15

30 Jul '15

Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 <-> server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514" Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?). RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error. So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on Transport: TCP IP: 212.126.160.92 Port: 6717 If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-)) Best regards, Joachim

3 4

Issue with ACK behind NAT
by Alberto Sagredo 30 Jul '15

30 Jul '15

I have Kamailio in a local network with RTPPROXY and NAT MANAGE I have seen an issue with ACK as show when Asterisk sends ACK to Kamailio, it sends to advertised Address instead to Kamailio IP address. What could i add to solve this? 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 dlg_manage(); } 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; } } BR

2 14

Re: [SR-Users] Memory Issue while accessing $rb (prakash)
by Surendra.P 30 Jul '15

30 Jul '15

Hi Prakash, Specify the more private memory with -M option, while running kamailio Like -M 64 :-) Regards @surendra

1 0

Planning to release v4.2.6
by Daniel-Constantin Mierla 30 Jul '15

30 Jul '15

Hello, I am planning to release v4.2.6 later this week, most likely on Thursday, July 30. If there is anything not reported that should be considered for this minor release of the branch 4.2, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Carrierroute and dispatcher
by Nelson Migliaro 29 Jul '15

29 Jul '15

Hello everybody, I am currently using Kamailio in order to separate traffic based on dialed number. Some traffic goes to several Asterisk in a load balance strategy and other traffic goes to other SIP devices. I use carrierroute module in order to route traffic based on destination. I use scan_prefix to classify traffic and rewrite_host to route the traffic. A destination number usually go to one of 4 asterisk server. I do this in order to load balance the traffic. In order to distribute traffic I use prob (0.25 for each destination). Using this configuration, I have to include on carrieroute table a row for each destination node. Since the amount of destination is huge, I have to repeat configuration in tables. It also complicates maintenance. I was thinking of using a combination of carrierroute and dispatcher module. Carrierroute for classify traffic, strip numbers, and rewrite destination host and dispatcher to distribute the traffic over asterisk. Using this configuration I have to indicate asterisk nodes once in dispatcher table once. Also in carrierroute table I have to indicate each prefix once. I was thinking in something like this. if(!cr_route("carrier", "domain", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); drop; } if(!ds_select_dst("1", "4")) { sl_send_reply("403", "Not allowed"); drop; } t_relay; Does it make sense, or am I complicating everything and there is a magic way to achieve this. Thank you.

2 1

Re: [SR-Users] Memory Issue while accessing $rb
by Daniel-Constantin Mierla 29 Jul '15

29 Jul '15

Hello, you have to increase the private memory -- see -M command line parameter -- as well as pv_buffer_size core parameter -- see core cookbook in the wiki. Cheers, Daniel On 29/07/15 13:37, prakash wrote: > > Hello, > > > > I am getting this Error: > > > > *ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory* > > *ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression > evaluation failed (881,17-881,29)* > > > > When I tried access *$rb* when sending large amount of data(*nearly > 4MB*) as a XML body. > > How can I sort out this issue? Is there anything to be configured > properly? > > > > NOTE: > > Version that I’m using kamailio 4.1.5 > > > > Thanks and Regards, > > Prakash P > > > > _______________________________________________ > 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Kamailio no audio
by goReeves Admin 29 Jul '15

29 Jul '15

Hi - I just installed Kamailio 4.3.0 on Ubuntu 14.04.2. I can connect using jitsi on windows and csipcimple on the android and do text messaging. However, when I try to do audio callis it seems that the mic is working but the audio out is not functioning. I used the tutorial to do the install so everything in the config files is pretty vanilla except for the database name, use and password. I also installed siremis and added my sip.xxxxx.com as a gateway. On the Ubuntu image I use VPN to a dedicated IP address on which all ports are open. I run web and cloud servers on this machine as well and they all work just fine. I am pretty new to this SIP stuff so it is probably my issue. Here is the network configuration: root@xxxxxxxx-VM:/var/www/siremis-4.2.0# ifconfig eth0 Link encap:Ethernet HWaddr xx:xx:xx:xx:xx:xx inet addr:192.168.x.x Bcast:192.168.x.255 Mask:255.255.255.0 inet6 addr: fe80::20c:29ff:fe84:447a/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1579455 errors:0 dropped:0 overruns:0 frame:0 TX packets:1444329 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:346579445 (346.5 MB) TX bytes:625010061 (625.0 MB) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:98232 errors:0 dropped:0 overruns:0 frame:0 TX packets:98232 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:262176392 (262.1 MB) TX bytes:262176392 (262.1 MB) ppp0 Link encap:Point-to-Point Protocol inet addr:xxx.xxx.xxx.xxx P-t-P:xxx.xxx.xxx.xxx Mask:255.255.255.255 UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1400 Metric:1 RX packets:1105547 errors:0 dropped:0 overruns:0 frame:0 TX packets:1228003 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:3 RX bytes:184558026 (184.5 MB) TX bytes:235399612 (235.3 MB) -- Web Administrator <mail:webadmin@goreeves.net> goreeves <https://www.goreeves.net>

2 2

Memory Issue while accessing $rb
by prakash 29 Jul '15

29 Jul '15

Hello, I am getting this Error: ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression evaluation failed (881,17-881,29) When I tried access $rb when sending large amount of data(nearly 4MB) as a XML body. How can I sort out this issue? Is there anything to be configured properly? NOTE: Version that I'm using kamailio 4.1.5 Thanks and Regards, Prakash P

1 0

double rr and $fs
by Asgaroth 29 Jul '15

29 Jul '15

Hi All, I have kamailio setup and listening on logical interfaces (for failover purposes) and therefore need to force the sending socket on initial messages. We are testing sips+tls at the moment and what we are trying to achive is that the TLS connection occurs from the UAC to the edge proxy (also responsible for the domain) and then force UDP for internal communications to services. What I am comming up against is that when I force the sending socket on the internal interface to UDP, the record route header still shows up as a "sips" request. Then, at the presense server, when generating the NOTIFY it attempts to send it to the proxy using TLS. My understanding was that the record route would set the uri something like "sip:internal_ip" when forcing the sending socket to be "udp:internal_ip". Is my understanding correct, or am I doing something wrong here? An example subscribe message follows, the top most record-route header, from my understanding, should read sip:internal_ip, not sips:internal_ip. Is this the expected behaviour of double rr with these types of messages? How could I ensure that the top most record-route would always be a sip uri (never sips). SUBSCRIBE sips:subscriber@domain.com:5061 SIP/2.0. Record-Route: <sips:proxy_internal_ip;r2=on;lr;ftag=594055226>. Record-Route: <sips:proxy_external_ip:5061;transport=tls;r2=on;lr;ftag=594055226>. Max-Breadth: 60. Via: SIP/2.0/UDP proxy_internal_ip;branch=z9hG4bK6c9.d7dc4801e872ce9fb5730f9e09f1889e.0;i=1. Via: SIP/2.0/TLS 172.16.0.121:5060;rport=33273;received=78.143.152.30;branch=z9hG4bK708ea7ba. Max-Forwards: 69. Call-ID: 17212ff4-4421321c8feafd63bf800080f0808080@KX-HDV230X. From: <sips:subscriber@domain.com>;tag=594055226. To: <sips:subscriber@domain.com>. CSeq: 2 SUBSCRIBE. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER. Accept: application/dialog-info+xml,application/rlmi+xml,multipart/related. Contact: <sips:subscriber@172.16.0.121:5060;alias=78.143.152.30~33273~3>. Expires: 3600. Event: dialog. Any thoughts, tips, tricks would be greatly appreciated. Cheers

2 2

Kamailio - Missing INVITEs
by Ali Taher 28 Jul '15

28 Jul '15

Hello, I connected Kamailio to a postgresql database (through sqlops module) and fetching result from a table containing around 300K records , where time processing of the single query is taking around 12 ms. I noticed that on huge load (around 200 - 300 calls per second) Kamailio is not processing all INVITE packets. For instance , I initialized a counter in Kamailio that increments on every INVITE , and on the other hand I counted the number of INVITEs found in Wireshark. I got a big difference (from 30K INVITES in Kamailio , I got 15K in Kamailio), though no errors were raised in kamailio log. How can I proceed in order to detect the problem? :( Any help would be appreciated. Regards, Ali

2 1

RR module - Fail detecting strict routing
by Antonio Reale 28 Jul '15

28 Jul '15

Hi all, I have the following scenario: U1 (caller) ---> P1 (192.168.0.245, kamailio 4.3, loose-router) ----> P2 ----> (192.168.0.101, strict router) ----> .... ----> U2 (called) When U2 answers the call, at P1 arrives the 200 OK with: Record-Route: <sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-28d2e> Record-Route: <sip:192.168.0.245;lr=on> The problem is that the ACK from U1 is forwarded from kamailio to P2, as if P2 is a loose router. P2 drops the ACK message. From kamailio logs I see: Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:88]: is_preloaded(): is_preloaded: No Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:783]: after_loose(): Topmost route URI: 'sip:192.168.0.245;lr=on' is me Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:869]: after_loose(): URI to be processed: 'sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-289ce' Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:878]: after_loose(): Next URI is a loose router Why after_loose function considers the next URI a loose router? The Route HF in the ACK with the URI of the next hop does not contain ;lr . It seems that the function is_strict fails detecting the strict router. Thanks. Regards. Antonio

2 7

CNXCC lock
by Kelvin Chua 28 Jul '15

28 Jul '15

after a call exhausted credits, the next call will fail with a timeout. found out it gets stuck in cnxcc_mod.c line 1137 static call_t *__alloc_new_call_by_money(credit_data_t *credit_data, struct sip_msg *msg, double credit, double cost_per_second, int initial_pulse, int final_pulse) { call_t *call = NULL; lock_get(&credit_data->lock); ... the next lines will never be executed Kelvin Chua

2 2

SIPCapture and Message Queues
by Grant Bagdasarian 27 Jul '15

27 Jul '15

Hello, Would it be possible to send the messages captured by the sipcapture module to a message queue instead of inserted directly into a database? Be it DMQ or SQL Server Service Broker Queues, etc. Or maybe another approach, executing a stored procedure instead of an insert statement? I couldn't find a way to accomplish this, but maybe someone has a brilliant idea how to accomplish any of the above. Regards, Grant

2 2

Problem Storing USRLOC Additional Attributes
by Luca Mularoni 27 Jul '15

27 Jul '15

Hi All, I'm using kamailio 4.2.5 (compiled from source and aligned up to commit [00a36ae07c587992d6486277ce2e9ae6c009685f]). In order to enrich users data collected during registering phase, I would like to store them in USRLOC Additional Attributes through the use of 'xavp_contact' structure, but I'm having problems storing them. Below relevant chunks of config: ... # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "xavp_contact", "ulvals") ... route[REGISTRAR] { ... # Just an example $xavp(ulvals=>A) = "A"; $xavp(ulvals[0]=>B) = "B"; $xavp(ulvals[0]=>C) = "C"; $xavp(ulvals[0]=>D) = "D"; ... if (!save("location")) sl_reply_error(); exit; } route[LOCATION] { ... if(!lookup("location") { # Handle Failure ... } # Print USRLOC Additional Attributes # Just an Example $var(i) = 0; $var(N) = $branch(count); while($var(i)<=$var(N)) { xlog("LOC - [$var(i)] - A [$xavp(ulvals[$var(i)]=>A)]\n"); xlog("LOC - [$var(i)] - B [$xavp(ulvals[$var(i)]=>B)]\n"); xlog("LOC - [$var(i)] - C [$xavp(ulvals[$var(i)]=>C)]\n"); xlog("LOC - [$var(i)] - D [$xavp(ulvals[$var(i)]=>D)]\n"); $var(i) = $var(i) + 1; } ... } And this is the relevant chunk of syslog when a local subscriber is found in 'location' table: ... LOC - [0] - A [A] LOC - [0] - B [<null>] LOC - [0] - C [<null>] LOC - [0] - D [D] ... Summing up, it seems that when xavp structure is stored in memory, just first and last elementsare inserted. In real config I'm using more than 4 attributes and behaviour is the same: just first and last are stored. I've tried also with modparam("usrloc", "db_mode", 0) but the problem remains. Where I'm wrong in using this structure? Or it could be a bug? Thank You in advance for your help Luca Mularoni

2 11

Check if user is registered after receiving reply
by Dmytro Bogovych 27 Jul '15

27 Jul '15

Greetings. I want to check if user is registered before kamailio will try to route reply to its user agent. IMHO it should be: onreply_route[MANAGE_REPLY] {

2 1

Re: [SR-Users] FROM number based on destination number
by Keith 27 Jul '15

27 Jul '15

Hi, I have a requirement to change the FROM number to a different number depending on the destination number. For example if I dial 01789 from 0800 I want to change the 0800 from number to a 01789 number. Ideally I want to be able to do a db lookup on the destination number to then set the from number. Thanks Keith

1 0

Changing acc_cdrs time and duration columns from varchar to int/float
by mayamatakeshi 27 Jul '15

27 Jul '15

Hello, I am planning to enable acc cdr in my kamailio 4.1 server. I am testing in my lab and it is working fine. However, when I check the table definition I see this: mysql> describe acc_cdrs; +------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | start_time | varchar(32) | NO | MUL | | | | end_time | varchar(32) | NO | | | | | duration | varchar(32) | NO | | | | +------------+------------------+------+-----+---------+----------------+ 4 rows in set (0.00 sec) The *_time and duration columns are defined as varchar. Because of this, i need to perform a cast when doing queries: select * from acc_cdrs where cast(start_time as unsigned) >= 1437120622; which would cause extra processing in the db server. So I experimented changing these columns to "int unsigned" and to "float unsigned" and it worked for both without problems (records properly added to acc_cdrs and no errors in kamailio.log): alter table acc_cdrs modify start_time int unsigned; alter table acc_cdrs modify end_time int unsigned; alter table acc_cdrs modify duration int unsigned; alter table acc_cdrs modify start_time float unsigned; alter table acc_cdrs modify end_time float unsigned; alter table acc_cdrs modify duration float unsigned; So I am thinking in doing this in my production server but I would like to know if there is any potential risk of doing this (as I am not sure why it was decided for these fields to be varchar instead of a numeric type). Regards, Takeshi

2 3

New SIP INVITE from UAS (new dialogue)
by Joao Alves 27 Jul '15

27 Jul '15

Hi, What would be the best way to generate and send an INVITE message generated by the Kamailio server. This would be for a new dialog session, UAS initiated (e.g. to make a click 2 call like service). I've tried the $uac_req(method)="INVITE"; uac_req_send(); but didn't work. I gather something would be needed first to open the dialogue before doing it but couldn't find it on the dialogue module. Thanks, Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

4 10

Issues sending SIP UPDATE inside an existing dialogue.
by Joao Alves 24 Jul '15

24 Jul '15

Hi, I need to send an SIP UPDATE inside an existing dialogue (in order to update the SDP offer). I've been using the following uac functions from the command file. $uac_req(method)="UPDATE"; $uac_req(ruri)=$shv(userAcontact); $uac_req(turi)=$tu; $uac_req(furi)=$fu; $uac_req(callid)=$ci; $uac_req(hdrs)="Content-Type: application/sdp\r\n"; $uac_req(body)=$var(sdp); # uac_restore_from(); uac_req_send(); As a result, the SIP UPDATE is being sent but since the original From Tag is not being used (i.e. and a new tag is generated), the SIP client (sipML5) does not recognize it as part of the existing dialogue. Attach is a pcap of the test made. Subsequently, I've also tried to recover the from using uac_restore_from() function but got the following an error message. DEBUG: uac [replace.c:511]: restore_uri(): getting 'vsf' Route param DEBUG: uac [replace.c:515]: restore_uri(): route param 'vsf' not found Thanks, Joao Alves Solution Architect, Unified Communications +351 214094660 (desk) +351 912783702 (mobile) AMDOCS | EMBRACE CHALLENGE EXPERIENCE SUCCESS This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

1 0

Installation issues
by ngn zone 24 Jul '15

24 Jul '15

>Hello all, I am new to Kamailio. I Tried installing Kamailio 4.0 and I had the following error. I will appreciate any help >ERROR: could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL'

2 1

dialog keepalive timeout - which parameter rules it?
by Andrey Utkin 24 Jul '15

24 Jul '15

What happens if dialog parties don't respond to keepalive requests sent by dialog module? It looks it is not coded in modules/dialog, I guess it can be set in t_uac_prepare() from modules/tm/uac.c. Please explain what timeout or lifetime parameters rule - at which delay the unanswered keepalive request sent by Kamailio within dialog triggers dialog termination. Thanks a lot. -- Andrey Utkin

2 2

dmq_usrloc replication and max_contacts
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where I have 2 devices registering with the same AOR, I have the registrar module's max_contacts parameter set to 1 and I use the 0x04 flag on the save function. When I use dmq_usrloc for replication, I can see 2 contacts registered for the AOR on the "client" nodes whereas the node where the actual registration took place only has 1 contact for the AOR. Is this expected behaviour of the dmq_usrloc module in terms of registration or have a come accross another bug? Thanks

2 5

"kamctl ul show"/xmlrpc ul.lookup AOR not found for valid registration
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where a device has successfully registered and its location saved/stored in memory. I can see all the stored information when I issue a "kamctl ul show" However, if I perform a kamctl ul show subscriber@domain, where the 'subscriber@domain' is copied/pasted from the output of the "kamctl ul show" command, it results in an AOR not found error. However, with other AOR's on the same registrar, the "kamctl ul show subscriber@domain" works. I checked the usernames for the failed commands, and I dont see anything obvious, they are a combination of lower/upper-case leters and digits. String length is around 24 characters for the username (the username table for the subscriber is 64 characters). I'm not sure what elese to check to see why the AOR lookup for these particular clients are failing when they are indeed registered. All other functions around the lookup seem to work, for example, a location lookup for these AOR's are working. I wonder if I have come accross an issue with the kamctl function and/or the equivilient xmlrpc ul.lookup method. Any thoughts on this would be greatly appreciated. Thanks

1 3

sending INVTE with Digest values
by Al S 24 Jul '15

24 Jul '15

I am registering my client to kamailo successfully. client --> Register --> kamailioclient <-- 401 with nonce value <-- kamailioclient --> Register with nonce and md5 response values --> kamailioclient <-- 200ok <-- kamailio However, when the client sends an invite with the same digest values, kamailio sends a 407 request for another challenge: client --> Invite with the same nonce and md5 response values --> kamailioclient <-- 407 <-- kamailio I am thinking the same digest values from register could be used for making calls. Thanks,Al

2 1

Dispatcher module number of calls
by Alberto Sagredo 23 Jul '15

23 Jul '15

Hi Hi have read documentation but it seems dispatcher does not keep how many calls has been dispatched or currently are in any of dispatcher destinations I have take a look to code on: http://lists.sip-router.org/pipermail/sr-users/2012-July/073919.html But it seems to use calls limit per user, I would need to establish a limit per dispatcher destination i order to do not send more calls. Any idea where to look for? I have take a look to DMQ module, that maybe i could to use to spread information of how many calls to every destination but do not know how to extract calls from kamailio that have been dispatched to this destination using disptcher module. Best Regards Alberto

3 5

Kazoo like load balancing
by Jurijs Ivolga 23 Jul '15

23 Jul '15

Hi, First of all I would like to mention, that I'm very new to Kamailio. :) My target is to create simple load-balancing similar what have Kazoo, basically 2 servers will share same domain name. I installed 2 Kamailio 4.3 servers and configured this 2 servers with one domain address. I used almost default config with: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_NAT #!define WITH_NATSIPPING Calls are going through if I register both clients to one server. But all my calls are failing, when clients are registered to different servers with error "404 Not Found". Maybe somebody can give me a hint how to make this work. Thank you! With kind regards, Jurijs

3 19

Re: [SR-Users] Kamailio mod_destroy() ERROR
by Ali Taher 23 Jul '15

23 Jul '15

Any help regarding the below would be appreciated. BR, Ali From: Ali Taher [mailto:ataher@vanrise.com] Sent: Wednesday, July 22, 2015 1:39 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Cc: 'Ali Taher~Vanrise Technical Support' <ataher(a)vanrise.com> Subject: Kamailio mod_destroy() ERROR Hello, I'm getting below error when starting Kamailio. ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Any advice ? Regards, Ali

2 1

Port 5061 Issue
by Ankhit Vivekananda 23 Jul '15

23 Jul '15

Hello, I have configured kamailio 4.0.4 long back. Now, I wanted to enable the usage of port 5061 with TLS. By doing so, the client(video camera) can place outgoing calls fine. But while receiving an incoming call, the message does not get sent out to the other client. Hence, the call fails. I can place and receive calls on port 5060 successfully without any issues. Can anyone please tell me why port 5061 is giving me a problem. Also, as a test, I set up kamailio on another machine(version 4.1) and 5061 works fine in all directions. It's only on the other machine that I have problems with. Appreciate any help! Thank you, Ankhit

2 1

htable replication between several kamailio's through dmq
by José Seabra 22 Jul '15

22 Jul '15

Hello there, I'm using the dmq and htable modules to limit the number of concurrent calls in several kamailio's servers. Every call that one kamailio server receives, kamailio increments a htable variable that is replicated to other kamailios through dmq, but if one of these kamailios fails or is restarted it looses the data from htable variable used to count the number of calls and it is not recovered again when kamailio starts again. Is there any way to recover the data from htable variable when kamailio starts, from another kamailio instances through dmq module? Best Regards -- Cumprimentos José Seabra

2 1

RTP Proxy to WAV Script conversion
by Alberto Sagredo 22 Jul '15

22 Jul '15

Using RTPBreak and sox im able to convert rtpproxy rtp files to wav. Later you could convert to mp3 if liked Just to share with list https://github.com/albersag/rtpproxy-utils Any comment is well appreciated Best Regards

2 1

dmq_is_from_node() issue
by Asgaroth 22 Jul '15

22 Jul '15

Hi All, I have an issue with the dmq_is_from_node() function, whereby it does not detect a node is on the bus if it has been removed from its "table" due to previously failed, i presume, ping requests. I have the following snippet in my main routing block: if(is_method("KDMQ")) { if (dmq_is_from_node()) { dmq_handle_message(); exit; } else { xlog("L_INFO", "KDMQ message recieved from node not on our bus!! ($si:$sp) Dropping."); exit; } } I have the following dmq module parameters enabled for dmq: modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:lb.domain.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) DMQ_ADDRESS is set to each of the addresses in the lb.domain.com for each local kamailio instance lb.domain.com resolves to the following addresses: 10.6.0.109 10.6.0.110 10.6.0.111 If I start up the 1st instance of kamailio and do not start the other 2 within the default ping timout then I get the expected log message of dmq removing the nodes from its table, for example: kamailio[26657]: ERROR: dmq [notification_peer.c:588]: notification_resp_callback_f(): deleting server sip:10.6.0.111:5060 because of failed request kamailio[26657]: ERROR: dmq [notification_peer.c:588]: notification_resp_callback_f(): deleting server sip:10.6.0.110:5060 because of failed request However, if I then start one, or both, of the other nodes, then the KDMQ requests are dropped due to my snippet above, I presume, because dmq_is_from_node() checks its "active" table. See messages below as an example of the above snippet in action: kamailio[26656]: INFO: <script>: KDMQ message recieved from node not on our bus!! (10.6.0.110:5060) Dropping. kamailio[26656]: INFO: <script>: KDMQ message recieved from node not on our bus!! (10.6.0.111:5060) Dropping. Is this expected behaviour from the dmq_is_from_node() function? I would have expected it to always check the addresses resolved in the notification address, or am I using this function in a manner not intended. Any suggestions/tips would be greatly appreciated. Thanks

3 5

RFC: about the self signed tls certificates
by Daniel-Constantin Mierla 22 Jul '15

22 Jul '15

Hello, (cross-posting as it impacts users as well) Currently we generate and install the TLS self-signed certificates when tls module is installed, including them in the debian packages (and I guess in rpms). Debian has a policy of reproducible builds, meaning that from same source tree snapshot the same binary packages should result. Even more important considering the impact on security, it would be better that the certificates are generated locally on the installation server, to be distinct. Right now, people relying on default installation config/certificates (and I guess there are many, at least in testing phase), are exposed to eavesdropping, because the private key is available in public packages. My proposal is to move generation of self signed certificates to kamctl. There can be a kamctl.tls file to be deployed by the tls package (same is done by kamctl.mysql, being part of mysql package), which should add a new group of commands, among them something like: kamctl tls generate-certificate The drawback is that before enabling tls and starting kamailio, one has to run the above command. We can document that in tls module readme and in kamailio.cfg in the comments related to WITH_TLS define. Anyone with comments, pros/cons? Other suggestions on how to address the reproducible builds as well as solve the security issue for the default installation? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

5 4

Kamailio mod_destroy() ERROR
by Ali Taher 22 Jul '15

22 Jul '15

Hello, I'm getting below error when starting Kamailio. ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Any advice ? Regards, Ali

1 0

replace From header username for AUTH
by Al S 22 Jul '15

22 Jul '15

Hi, I was wondering how can I replace From header username to perform the following authentication check with the new value: route[AUTH]......if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "1"); exit;}... Thanks,Al

2 1

kamailio 4.3v partition column name is reserved word in mysq ndbcluster engine
by José Seabra 21 Jul '15

21 Jul '15

Hello Daniel, After update my kamailio to version 4.3 and change the table location as required in migration process, kamailio start giving some errors: [root@ccp1 kamailio]# 35(7637) ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'partition' in 'where clause' 35(7637) ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query that's is because the column name "partition" is reserved word ( http://dev.mysql.com/doc/refman/5.6/en/keywords.html) in mysql ndbcluster engine, so what i did to fix this issue was add a new modparam in modules/usrloc/ul_mod.c file: {"partition_col", PARAM_STR, &partition_col}, with this i can configure what name i want for this column in kamailio module configuration and configure a diferent column name in table location(ex: modparam("usrloc", "partition_col", "partitionusrloc") ), Can you please check out if what i did is the correct way to fix it, in order to avoid damage something else. Best regards -- Cumprimentos José Seabra

2 6

Compiling only one module
by Loic Chabert 21 Jul '15

21 Jul '15

Hello, I'm writing a new module for influxdb stats. I've wrote my code and now i need to compile it. How can i compile it ? I don't want to compile all modules, only this one. Thanks in advance, Regards.

2 3

usage of t_flush_flags() in event_route
by Andrew Pogrebennyk 20 Jul '15

20 Jul '15

Hi Daniel and others, I'm having a problem with acc module if I'm using the event_route/ branch-failure: say, the call comes from the app server and goes to the registered user. We arm the the failure route and per-branch failure route for the 302 redirect from the UA. We explicitly reset the accounting flags because we don't want to account the calls from the app server. The transaction is created implicitly by the t_relay(). Now if the UA responds with the 302 response and we are going to process that, we want to create an acc record for the new target from 302 message because this call may incur additional costs. I'm setting the accounting flags and calling t_flush_flags() but that doesn't work (no accounting record). Any idea if I'm doing something wrong or maybe there's a bug when changing the flags and then calling t_flush_flags from the event_route? Here are the module parameters: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "db_url", PAIR_URL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "cdr_log_enable", 0) FTR, we tried 4.1.6 and 4.1.8. And here is the event route (with flags defined like this: flags FLAG_ACC_DB:1, FLAG_ACC_MISSED:2, FLAG_ACC_FAILED:3, ...): event_route[tm:branch-failure:redirect] { route(ROUTE_STOP_RTPPROXY_BRANCH); if($T_rpl($rs) == 301 || $T_rpl($rs) == 302) { # initialise variables when entering failure route route(ROUTE_INITVARS); # these need to be avps because we need it in reply/failure-route $(avp(s:from_faxserver)[*]) = 0; $(avp(s:to_faxserver)[*]) = 0; $(avp(s:cf_from_pstn)[*]) = 0; $(avp(s:from_pstn)[*]) = 0; $(avp(s:proxylu_from_pstn)[*]) = 0; $(avp(s:lcr_flags)[*]) = 0; $(avp(s:em_call)[*]) = 0; $(avp(s:from_pbx)[*]) = 0; $(avp(s:p_to_device)[*]) = 0; $(avp(s:p_to_group)[*]) = 0; $(avp(s:is_primary)[*]) = 0; # now let's process a 30x $(avp(s:acc_state)[*]) = "cfb"; $(avp(s:orig_acc_caller_user)[*]) = $avp(s:acc_caller_user); $(avp(s:orig_acc_caller_domain)[*]) = $avp(s:acc_caller_domain); $(avp(s:acc_caller_user)[*]) = $avp(s:acc_callee_user); $(avp(s:acc_caller_domain)[*]) = $avp(s:acc_callee_domain); $(avp(s:caller_uuid)[*]) = $avp(s:callee_uuid); $(avp(s:callee_uuid)[*]) = $null; # the $var(no_acc) is 0 at this point but the flags may have been reset # if this is a call from PBX user - we do want accounting for the 302 redirect if(isflagset(FLAG_ACC_DB)) { xlog("L_NOTICE", "++++++ ACC flag is set - [% logreq -%]\n"); } else { xlog("L_NOTICE", "------ ACC flag is NOT set - [% logreq -%]\n"); } setflag(FLAG_ACC_FAILED); setflag(FLAG_ACC_DB); t_flush_flags(); # get last URI from destination-set and set it as R-URI $var(contact) = $T_rpl($ct); $var(contact) = $(var(contact){nameaddr.uri}); if($var(contact) == 0 || $var(contact) == $null) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [% logreq -%]\n"); acc_db_request("480", "acc"); $var(announce_handle) = "callee_tmp_unavailable"; $var(announce_set) = $xavp(callee_real_prefs[0]=>sound_set); $(avp(s:announce_code)[*]) = 480; $(avp(s:announce_reason)[*]) = "Temporarily Unavailable"; route(ROUTE_EARLY_REJECT); } $ru = $var(contact); xlog("L_NOTICE", "Redirect from UAC intercepted - [% logreq -%]\n"); $(avp(s:forwarder_cli_userprov)[*]) = $T_rpl($tU); $(avp(s:forwarder_domain_userprov)[*]) = $T_rpl($td); $var(forward) = 1; $var(redirected_forward) = 1; route(ROUTE_LOAD_CALLER_PREF); route(ROUTE_FIND_CALLEE); } } Thanks. Andrew

2 5

File Transfer
by Abdul Hakeem 20 Jul '15

20 Jul '15

Hello, Is there a support for file transfer between registered SIP clients, If not, how can this be achieved ? Regards, Abdul Hakeem

1 0

Kamailio v4.3.1 Released
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, Kamailio SIP Server v4.3.1 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.3.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.1. For more details about version 4.3.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-3-1-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

Missing BYE at the far end
by Kliment Toshkov, Netfinity JSC 20 Jul '15

20 Jul '15

Dear colleagues, We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE. Current setup is the following: PEER IP > KAMAILIO 4.2.2 > (Asterisk PBX 1, 2, 3, …, N) We provision only termination services, which is: 1) peer calls us 2) Kamailio chooses destination using dispatcher module 3) forwards the call to it. Attached is captured trace. Clearly BYE is retransmitted a few times, then request for UPDATE is received but it is not possible to answer it because the call is already hung up (as far as I understand). Kamailio routing file is available upon request. Thank you in advance! Климент Тошков Изпълнителен директор 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

2 1

Planing to release v4.3.1
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, I am considering to release the first patch version in 4.3 series by beginning of next week (Monday or Tuesday, July 20 or 21). If you are aware of issues not reported yet, then open an item on the tracker to be taken care: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
by Ben Fitzgerald 20 Jul '15

20 Jul '15

Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: <sip:40081@kamailio:5060> Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER ========================= sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

3 7

Double Record-Route to subscriber with multiple locations on different SIP transports
by Andrey Utkin 17 Jul '15

17 Jul '15

Hi! We found that we need double Record-Route when we call from one SIP transport to another (let's say, TCP -> TLS), because without |transport=" hint in Record-Route, mobile Linphone app sends messages (INVITE reply, ACK, BYE) via UDP, even if configured for another transport. record_route_advertised_address() is told to add double RR if needed, automatically, so we use it. But the subscriber (the receiver of call) can have multiple registered locations, let's say, one connected via TCP, and another via TLS. So we need to handle each call branch added by lookup(). But adding record_route_advertised_address() ONLY to branch_route attached by t_on_branch() doesn't execute if there's only one location. Having record_route_advertised_address() just in routing script after lookup() seems wrong because it seems unable to edit it for other branches (I don't remember if I have tested this and have proven a failure, or this is just my speculation). So we have added it in both places - right under lookup(), and in branch_route. But in this case we have an error logged: ERROR: rr [rr_mod.c:337]: w_record_route_advertised_address(): Double attempt to record-route Could anyone explain what would be the most correct solution? What is intended behaviour of branch_route? Is it omitted from execution on "primary" branch of call? Is there any parameter which could help? Thanks in advance. -- Andrey Utkin

3 7

Dialogue restore after initial processing
by Joao Alves 17 Jul '15

17 Jul '15

Hi, I'm trying to restore a SIP session dialogue context and found dlg_get(callid, ftag, ttag) function for that. However, I cannot access the To tag. I noticed on the 180 RINGING there was already an self-generated included on this response, but I also could not access its value. When receiving the SIP INVITE request I used the dlg_manage() and the context was initially stored. if(is_known_dlg()) { xlog("Request $rm from $ci is in-dialog\n"); xlog(">>From tag is $ft and To tag is $tt\n"); # Needed to later restore the dialog $shv(ftag)=$ft; $shv(ttag)=$tt; } 15(49100) ERROR: <script>: Request INVITE from 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ is in-dialog 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1021 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From tag is 8fd06474 and To tag is <null> Since there was a self-generated one (after the 180 RINGING being sent), I rechecked the $tt and $dlg(to_tag) values and both still returned null values. 15(49100) DEBUG: tm [t_lookup.c:1083]: t_check_msg(): DEBUG: t_check_msg: T already found! 15(49100) DEBUG: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (49100 (15)), reusing fd 15(49100) DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 15(49100) DEBUG: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7effdf7da860 n=396 fd=5 15(49100) DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= SIP/2.0 180 ringing Via: SIP/2.0/TCP 192.168.1.5:58751;branch=z9hG4bK-524287-1---a64e0a42199ee419;rport=61134;received=85.138.200.8 To: <sip:1001@decasa.website>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f6a3 From: "1000"<sip:1000@decasa.website>;tag=8fd06474 Call-ID: 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ CSeq: 1 INVITE Server: kamailio (4.3.0 (x86_64/linux)) Content-Length: 0 ..., shmem=0x7effdf7e1e00: SIP/2.0 180 ringingght(): DEBUG: reply sent out. buf=0x7effec023bf8: SIP/2.0 180 ringing 15(49100) DEBUG: tm [t_reply.c:657]: _reply_light(): DEBUG: _reply_light: finished 15(49100) DEBUG: sl [sl.c:272]: send_reply(): reply in stateful mode (tm) 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1031 a=25 n=xlog 15(49100) DEBUG: dialog [dlg_hash.c:640]: dlg_lookup(): ref dlg 0x7effdf7e1bf0 with 1 -> 2 15(49100) DEBUG: dialog [dlg_hash.c:642]: dlg_lookup(): dialog id=5769 found on entry 128 15(49100) DEBUG: dialog [dlg_hash.c:858]: dlg_unref(): unref dlg 0x7effdf7e1bf0 with 1 -> 1 15(49100) ERROR: <script>: Dialog To Tag: <null> <== xlog("Dialog To Tag: $dlg(to_tag)\n"); 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1033 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From2 tag is 8fd06474 and To tag is <null> Many thanks, Joao Alves Solution Architect, Unified Communications This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

2 3

Configuring Kamailio as SIP Proxy
by Djamel Bahamid 16 Jul '15

16 Jul '15

Hi, I have terminals of video conference which are in public IP ( *DMZ*). I installed kamailio (4.2) which works correctly, it is also in IP pulique (*DMZ*). I wish configured the /kamailio.cfg/ file, so that kamailio works in *proxy SIP*. And declared my CODEC of visoconférence in URI, example: 'DR London(a)domaine.fr'. In what part of the file ( *kamailio.cfg* ) do I have to make these modifications ? This to avoid the flooding. How possibly write the rule of corresponding ACL. Thank you for your help. Best Regards,

3 4

interconnect two Kamailio server
by Kai Ohnacker 16 Jul '15

16 Jul '15

Hello, I have a setup with 2 Kamailio server and their own database. Is it possible to connect these 2 Kamailio server, that the client from the one can make a call to the other? If yes, how could this be realized? Many Thanks. Cheers, Karl --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com

2 1

Error forwarding to SCSCF
by Jan Devai 16 Jul '15

16 Jul '15

*Hi Experts,* *I would like to ask you if some of you have already come across this problem.* *We are integrating Kamailio all in one Solution (P/I/S-CSCF having all same IP only communicating between each other on different ports) taken from Carsten Bock example against an external HSS.* *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF --------------------------Sip Register--------------------------> ----------Sip Register---------> <---------------------------------UAR-------------------------------- ---------------------------------UAA--------------------------------> --------------SIP Register-------> *The last step "SIP Register" from I-CSCF to S-CSCF is not happenning for some reason.* *The Received AVP in UAA is "First Time Registration 2001 code".* *Now we do not know whether this UAA answer is OK, or some other URI is badly defined which could explain why I-CSCF has trouble contacting S-CSCF.If someone have some logs, traces which could be used as a reference for this scenario would be greatly appreciated.* *Find theses messages we get in the shown logs below:* *Thank you* *JanThis file /rtc/kamailio_icscf/kamailio.cfg is a function which sends message to SCSCF, see below:* route[REG_UAR_REPLY] { #xlog("$$avp(s:uaa_return_code) = $avp(s:uaa_return_code)\n"); #this is async so to know status we have to check the reply avp switch ($avp(s:uaa_return_code)){ case 1: #success if (I_scscf_select("0")){ t_on_failure("register_failure"); t_on_reply("register_reply"); #now relay to appropriate SCSCF if (!t_relay()) { t_reply("500", "Error forwarding to SCSCF"); } } else {#select failed I_scscf_drop(); t_reply("500", "Server error on SCSCF Select (UAR)"); } break; case -1: #failure xlog("L_ERR", "UAR failure - error response sent from module\n"); break; case -2: #error xlog("L_ERR", "UAR error - sending error response now\n"); t_reply("500", "UAR failed"); break; default: xlog("L_ERR", "Unknown return code from UAR, value is [$avp(s:uaa_return_code)]\n"); t_reply("500", "Unknown response code from UAR"); break; } } *Kamailio is getting an answer: "t_reply("500", "Error forwarding to SCSCF");"* *which means that th function I_scscf_select("0") is fine but the other function called "t_relay", is the one generating the error "Error forwarding to SCSCF".* *I-CSCF is not able to send the SIP message to S-CSCF, see this log below:* Jul 16 14:59:22 cw-aio /usr/sbin/kamailio[31604]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31599]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: scscf.epc.mnc004.mcc324.3gppnetwork.org Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches

1 0

Purpose of table location_attrs
by mayamatakeshi 16 Jul '15

16 Jul '15

Hello, I need to prepare two kamailio instances accessing the same mysql db. They will write to different location tables so i use function calls like these: Server1: save("k1_location"); lookup("k1_location"); Server2: save("k2_location"); lookup("k2_location"); This is working fine. However, checking db, I can see there is also a table called location_attrs and I am not sure what is its purpose (is this documented?). Anyway, I suspect I should also set a separate copy of it for each kamailio instance, but I am not sure how to do it. Regards, Takeshi

2 2

Error: make modules include_modules ="db_mysql" cfg
by kumar m 16 Jul '15

16 Jul '15

I am a newbie trying out kamailio *kamailio version: *4.3.0 *Operating system:* CentOS 7 Following the instructions provided at http://www.kamailio.org/wiki/install/4.3.x/git While I was trying to include the module "db_mysql", I get the following error [root@localhost kamailio]# make modules include_modules="db_mysql dialplan app_mono" cfg Makefile:56: modules.lst: No such file or directory target architecture <x86_64>, host architecture <x86_64> saving modules list... target architecture <x86_64>, host architecture <x86_64> which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) make --no-print-directory -wC ../.. cfg-defs target architecture <x86_64>, host architecture <x86_64> making config... which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) CC (gcc) [M db_mysql.so] km_dbase.o km_dbase.c:36:25: fatal error: mysql/mysql.h: No such file or directory #include <mysql/mysql.h> ^ compilation terminated. make[1]: *** [km_dbase.o] Error 1 make: *** [modules] Error 1 Please help me in fixing the error

2 1

Unclear documentation for "priority" field in Dispatcher module
by Sven Neuhaus 16 Jul '15

16 Jul '15

Hello, the documentation doesn't clearly state if a larger value for the priority field means that it gets used first before entries with smaller priority values are used - or vice versa. It currently states: “8” - select destination sorted by priority attribute value (serial forking ordered by priority). and priority: sets the priority in destination list (based on it is done the initial ordering inside the set) This is just ambiguous. It should clearly say wether the list is processed in ascending or descending order for the priority value when using algorithm 8 (serial forking aka failover). Regards, -Sven Neuhaus

2 1

Issue handling SRTP and RTP with rtpproxy and rtpengine
by Alberto Sagredo 15 Jul '15

15 Jul '15

Hi Im using an Asterisk cluster behind Kamailio and some services are running on Asterisk. Traffic always must go to Asterisk so RTPPROXY and RTPNGINE do the work and send traffic to Asterisk and traffic from Asterisk go to RTP Proxy I have been able to get rtpproxy working and later tried RTPengie to be able to manage SRTP I have been able to make SRTP To RTP to Asterisk Im able to call between RTP using rtpproxy as recording is a must and rtpengine does not record yet. But im not able to call between SRTP extensions, i understand also SRTP to RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine trasncode) I have this code on NATMANAGE but i do not know what could be wrong to handle SRTP Any idea? Thanks in advance if(src_ip==ASTERISK1_LAN || src_ip==ASTERISK2_LAN ){ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-kamtoast"); set_rtp_proxy_set("1"); rtpproxy_manage("rwei"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-kamtoast"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove RTP AVP"); } } } else{ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-asttokam"); set_rtp_proxy_set("1"); rtpproxy_manage("rwie"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-asttokam"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove "); } } } If you need any more info let me know.

4 12

Missing BYE at the far end
by Kliment Toshkov, Netfinity JSC 15 Jul '15

15 Jul '15

Dear colleagues, We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE. Current setup is the following: PEER IP > KAMAILIO 4.2.2 > (Asterisk PBX 1, 2, 3, …, N) We provision only termination services, which is: 1) peer calls us 2) Kamailio chooses destination using dispatcher module 3) forwards the call to it. Attached is captured trace. Clearly BYE is retransmitted a few times, then request for UPDATE is received but it is not possible to answer it because the call is already hung up (as far as I understand). Kamailio routing file is available upon request. Thank you in advance! Климент Тошков Изпълнителен директор 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

1 0

ISUP decode and store to be able to append_body_part() later
by andre second 15 Jul '15

15 Jul '15

Hi, I am trying to extract and save ISUP from SIP-I packet for a later use. The problem is that data is stored in binary format and needs decoding before saving to a variable. Can Kamailio perform any decoding so that later I can use append_body_part() to append it back to a multipart message? Thanks,Andrei

1 0

License for SIP-Router server when used as part of a commercial product development
by Mahesh Kumaraguru 15 Jul '15

15 Jul '15

We wish to use SIP-Router server for our commercial product development. We plan to develop SIP clients for Android, iOS, Windows phone and PC browser to have voice and video calls. Please let us know the licensing terms under which SIP-Router server is available, especially in the context of us using it for our commercial product. Many Thanks and Best Regards, Mahesh Kumaraguru +(91)-942-417-4471 +(91)-835-782-9220 mahesh.kumaraguru(a)vectramind.com<mailto:mahesh.kumaraguru@vectramind.com> http://www.vectramind.com/

1 0

License for iptel SIP server when used as part of a commercial product development
by Mahesh Kumaraguru 15 Jul '15

15 Jul '15

We wish to use iptel SIP server for our commercial product development. We plan to develop SIP clients for Android, iOS, Windows phone and PC browser to have voice and video calls. Please let us know the licensing terms under which iptel SIP server is available, especially in the context of us using it for our commercial product. Many Thanks and Best Regards, Mahesh Kumaraguru +(91)-942-417-4471 +(91)-835-782-9220 mahesh.kumaraguru(a)vectramind.com<mailto:mahesh.kumaraguru@vectramind.com> http://www.vectramind.com/

1 0

Issues with Yealink SDP and sdp_with_transport()
by Alberto Sagredo 14 Jul '15

14 Jul '15

Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy supports recording and rtpengine does not yet. So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones all worked fine, but when configuring Optional or Compulsory SRTP in Yealink it seems to do not detect i have seen that crypto lines are not in the final SDP but do not know if thats the reason Did you have a similar issue with Yealink? If i could get traces in anyway to help let me know. BR Alberto INVITE sip:212@10.0.1.34:15060 SIP/2.0. Record-Route: <sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes>. Record-Route: <sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes>. Via: SIP/2.0/UDP x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1. Via: SIP/2.0/TLS 10.0.1.111:11891 ;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360. From: "214" <sip:214@1x.x.x.x:8001>;tag=1072578853. To: <sip:212@x.x.x.x:8001>. Call-ID: 0_1310998066(a)10.0.1.111. CSeq: 2 INVITE. Contact: <sip:214@83.x.x.x:11891;transport=TLS>. Content-Type: application/sdp. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. Max-Forwards: 69. User-Agent: Yealink SIP-T21P_E2 52.80.0.3. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 553. . v=0. o=- 20143 20143 IN IP4 x.x.x.x. s=SDP data. c=IN IP4 x.x.x.x t=0 0. m=audio 8530 RTP/AVP 0 8 18 101. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3. a=crypto:3 F8_128_HMAC_SHA1_80 inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=nortpproxy:yes

2 2

Kamailio Load balancing Across Multiple Asterisk Servers -Chanspy
by Jonathan Hunter 14 Jul '15

14 Jul '15

Hi guys, I have a kamailio load balancing requests across 2 Asterisk Servers, and all is working well. Only problem now is when utilizing feature codes to look to listen into calls using the chanspy application I am having issues. I am passing calls to *2222 into Asterisk, where an IVR answers the call, then sends an INVITE back to kamailio to check to see if the extension I want to listen into is in an active call (hash table), and the call will then be forwarded to the Asterisk in question and chanspy applied so it can be listened to. The problem I have is when I send an INVITE from Asterisk Server, and then kamailio sends an INVITE back to the same Asterisk Server I get the following failure in asterisk console; Not accepting call completion offers from call-forward recipient I appreciate its potentially more an Asterisk question but I wondered if anyone had experienced the same issue? I have setup a completely load balanced environment so dont want to use Primary/Secondary Server routing. Thanks Jon

2 4

Nat Detect does not work with GS Phones
by Alberto Sagredo 14 Jul '15

14 Jul '15

I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on this phones Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind nat so NATMANAGE is not called Here its some trace. ANy clue how to handle this phones if they activate STUN for example U 80.26.x.x:52768 -> 192.168.0.170:8002 INVITE sip:2@x.x.x.x:8002 SIP/2.0. Via: SIP/2.0/UDP 80.26.x.x:52768;branch=z9hG4bK358742535;rport. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1478444456. To: <sip:2@x.x.x.x:8002>. Call-ID: 244257786-52768-56(a)IA.CG.BIE.BCH. CSeq: 550 INVITE. Contact: "Anonymous" <sip:212@80.26.x.x:52768>. X-Grandstream-PBX: true. Max-Forwards: 70. User-Agent: Grandstream GXP2140 1.0.4.23. Privacy: id. P-Preferred-Identity: <sip:212@x.x.x.x:8002>. Supported: replaces, path, timer. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 335. . v=0. o=212 8000 8000 IN IP4 80.26.x.x. s=SIP Call. c=IN IP4 80.26.x.x. t=0 0. m=audio 55422 RTP/AVP 0 8 18 9 2 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:9 G722/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # 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; }

4 6

Possible problem with multiple threads handling same SIP message
by Dirk Teurlings - SIGNET B.V. 14 Jul '15

14 Jul '15

Hello, Running kamailio 4.2.5 for a few weeks now, coming from 4.2.3. Since the upgrade we increased the logging substantially in our routing to get more insight into what is going on. We noticed that in a few cases the log showed a SIP message was being threated by multiple forks, each considering if it was a non-existing call. e.g. Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions As you can see this shows different threads handling the exact same callid (printed by $ci). And both run the same checks in our routing plan. The main problem is, I'm not able to reproduce this on the testplatform, but only see it happen on the main systems. Also, it doesn't happen that often, in this case there are about maybe 10 similar calls like this over the run of a few weeks. This case show the call being handled by 2 threads, but we've even seen with one call it being handled by 4 threads, all with the same $ci and the same messages, just different PIDs. I've checked and made sure the routing isn't recursively calling itself. But even if that was true, I would imagine the same PID for all the log entries. Running production in debug mode isn't an option, as it will interfere too much with the day-to-day business. So I'm hoping to get a few pointers here maybe, I'm happy to look at the source code more in depth, I just need a little help in the right direction to not get lost in other code. Any help is much appreciated. Cheers, Dirk

2 2

Re: [SR-Users] code coverage in kamailio using gcov
by Daniel-Constantin Mierla 14 Jul '15

14 Jul '15

Hello, On 09/07/15 12:54, Surendra Pullaiah wrote: > > Hello > > > > I am trying to implement code coverage tool in > kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and > it got compiled successfully. > > My problem is when I am running kamailio it is saying > lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. > > > > Can anyone help me on this. > Maybe those defines and libs need to be added to internal libraries -- see the Makefiles inside the folders of lib/ directory in kamailio source tree. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Kamailio as pass through proxy
by kai.ohnacker＠cbc.de 14 Jul '15

14 Jul '15

Hello, I try to setup a SIP infrastructure with Kamailio server in private network and a pass through Kamailio proxy in the DMZ for NAT handling. (-> TEST.jpg) The pass through proxy uses a new routing algorithm and this works fine for external calls to internal calls. (->Kamailio.cfg.txt) The logic behind is if a packet on external interface arrives, the proxy change the URI and forward the packet to the internal SIP server. Otherwise, the proxy shall forward the packet to external interface if a packet arrives at the internal interface. Because the SIP server has a fix IP, it's easy to route to this server, but the way back is more complicated and I think my algorithm is not working. But the first problem is the routing behavior from the private server if I try to make a call from intern to extern. The server sends the request to the external IP address from the client and the IP address from the pass through Kamailio proxy is not considered. 8(2720) ERROR: <core> [udp_server.c:576]: udp_send(): sendto(sock,0x7f4724dd9238,1328,0,192.168.1.20:5060,16): Network is unreachable(101) 8(2720) ERROR: tm [../../forward.h:208]: msg_send(): udp_send failed 8(2720) WARNING: tm [t_fwd.c:1610]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=553 a=24 n=sl_reply_error 8(2720) ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=555 a=2 n=exit 7(2718) exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=474 a=5 n=route To change the IP address which is used from the private server as the destination address, I added the following lines to the Kamailio.cfg. modparam("registrar", "use_path",1) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received",1) modparam("path", "use_received", 1) But the routing behavior has not changed. (perhaps this configure was not the right) Is there a way to configure the private server that the last IP address from the record-route header is used? Cheers, Kai

2 1

Removing headers in a failure route on 3xx redirect
by Daniel Tryba 13 Jul '15

13 Jul '15

I'm adding and removing headers in the standard route[RELAY], but if there is a redirection involved headers aren't removed. route[RELAY] { #stdstuff route(ADDCHECKSUM); # 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; } route[ADDCHECKSUM] { if($avp(dst_accountcode) || $avp(src_accountcode)) { #some magic remove_hf("X-rand"); remove_hf("X-csum"); append_hf("X-rand: $var(rand)\r\n"); append_hf("X-csum: $var(checksum)\r\n"); } return; } This works well unless the destination is a redirecting sip server. failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if (t_check_status("3[0-9][0-9]")) { #... route(RELAY); exit; } -user calls destination. -dispatcher sets destination is send to redirectserver and routes to RELAY, addchecksums add some headers called from relay. -redirectserver replies 302. -failure route triggers and ends with a route to RELAY. Any existing old values should be removed and the newer ones added. But end the result is that I have two sets op X-rand/csum headers for the INVITE to the redirection contact. How do I get rid of every X-csum/rand header except the last added ones? Version tested: 4.2.5.

2 2

Pipelimit
by Fred Posner 13 Jul '15

13 Jul '15

I know in ratelimit, you can define a method within queues... for example, modparam("ratelimit", "queue", "3:INVITE") to match on INVITE method. With pipelimit, is the method called from the config, such as: if(is_method("INVITE")) { # Checking limit on INVITE if (!pl_check("$au", "traildrop", "$var(limita)")) { pl_drop(); exit; } } if(is_method("REGISTER")) { # Checking limit on REGISTER if (!pl_check("$au", "traildrop", "$var(limitb)")) { pl_drop(); exit; } } -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

2 1

msilo delete rows
by Evgeny Zhavoronkov 13 Jul '15

13 Jul '15

Hi All, I'm fighting with msilo module now. I recive an offline message, but msilo gets 408 status and think that a message wasn't sent succesfully. According to the log, the error somewhere in t_relay, but I couldn't find where exactly. Therefore, msilo doesn't delete rows from silo table in database and everytime the user reregisters he gets offline messages again and again. -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

1 0

Does Kamailio support Cx interface Diameter over SCTP, somebody tested it?
by Jan Devai 13 Jul '15

13 Jul '15

Hi Team, Does anybody know if Kamailio support Cx interface Diameter over SCTP, somebody tested it? Thank you Jan

2 1

working config
by lifayk 13 Jul '15

13 Jul '15

Hello All! Can somebody share a working/example config with biderectional nathelper, tcp transport and push notifications (tsilo) for Kamailio >=4.3? Thanks in advance! -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

2 1

sip: uri to tel: uri
by manuel espinoza 13 Jul '15

13 Jul '15

Hi, The arch looks like this SBC1 <-------- SIP-Router (Kamailio) <------------- Asterisk Asterisk sends an invite to kamailio using "sip:" as request uri scheme, but SBC1 expects that invite to arrive using tel:, is there a way to change this in kamailio and forward that invite? Something like: $ru = "tel: +" + $rU; Thanks!!!

2 1

sctp on Centos 7
by micunek＠gmail.com 13 Jul '15

13 Jul '15

Hi, I would like to enable and use sctp in kamailio but I do not know exactly how to do it. It seems that documentation is a little bit inconsistent. 1. I have Centos 7 and kamailio installed from repositories. 2. I set loadmodule "sctp" in kamailio.cfg 3. I got this error: Jul 9 08:08:54 kamailio: : sctp [sctp_mod.c:99]: mod_register(): sctp core support not enabled Jul 9 08:08:54 kamailio: ERROR: <core> [sr_module.c:600]: load_module(): /usr/lib64/kamailio/modules/sctp.so: mod_register failed Jul 9 08:08:54 cw-aio kamailio: : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio.cfg, line 230, column 12-17: failed to load module 4. I checked mod_register() function and found out that USE_SCTP should be set. 5. kamailio -V did not show USE_SCTP 6. Next I tried to compile kamailio sources from 4.3 branch make include_modules="db_mysql tls sctp" SCTP=1 cfg make all 7. Still the same problem "mod_register(): sctp core support not enabled" 8. Then I added USE_SCTP to C_DEFS and compile again. This attempt was successfull. I was able to see USE_SCTP in kamailio -V output. Questions: 1. What is the right procedure to enable sctp in kamailio? 2. Why the -DUSE_SCTP was not used when SCTP=1 was used. According to documentation: "The core Makefile variable SCTP must be set to 1 (which is by default set to 1 in Makefile.defs) and sources compiled with -DUSE_SCTP (automatically set when SCTP=1). In other words, if core Makefiles are not changed and SCTP variable is not overwritten from command line, then the SCTP support in core is enabled." Regards, Peter

2 1

nat_uac_test question
by Kristian Kielhofner 13 Jul '15

13 Jul '15

Hello, I'm testing some basic flags with nat_uac_test. I have the following (rudimentary) script: route[NATDETECT] { #!ifdef WITH_NAT if (has_body("application/sdp")) { if (nat_uac_test("18")) { xlog("SCRIPT: SDP NAT Detected\n"); setflag(6); } } else { if (nat_uac_test("2")) { xlog("SCRIPT: Other NAT Detected\n"); setflag(6); } } if (isflagset(6)) { xlog("SCRIPT: NAT detected\n"); force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } When presented with this REGISTER message (ngrep output, so first line is IP source address and port): U 192.168.113.17:6060 -> 192.168.113.11:5060 REGISTER sip:192.168.113.11;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport. From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40. To: <sip:user_1@192.168.113.11>. CSeq: 10 REGISTER. Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40. Max-Forwards: 70. User-Agent: FreeSWITCH. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600. Content-Length: 0. . U 192.168.113.11:5060 -> 192.168.113.17:6060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport=6060;received=192.168.113.17. From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40. To: <sip:user_1@192.168.113.11>;tag=b27e1a1d33761e85846fc98f5f3a7e58.1045. CSeq: 10 REGISTER. Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40. Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600;received="sip:192.168.113.17:6060". Server: Kamailio. Content-Length: 0. . Log output: Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>: SCRIPT: Other NAT Detected Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>: SCRIPT: NAT detected Log output, the presence of the added "received" parameter in the returned Contact, etc show that NAT was detected. However, the IP source and port match the one presented in the Via header. What am I missing here? Thanks! -- Kristian Kielhofner

2 1

pua_publish MWI message via xmlrpc
by Asgaroth 10 Jul '15

10 Jul '15

Hi All, I am having difficulty getting pua_publish to issue a NOTIFY when I send an XMLRPC command to generate a PUBLISH into the presence server. I have been reading the pua_mi module docs, but I cannot see where I am going wrong with the following xmlrpc command, please would someone point me in the correct direction. I have an XML file which is the basis of the XMLRPC request via curl, I have attached it to this email for you to view, please would you take a look and let me know where I am going wrong, I suspect it is my message body as the pua module complains about the message body being NULL: Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [add_events.c:125]: pres_process_body(): while parsing xml memory Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [send_publish.c:650]: send_publish(): while processing body Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [send_publish.c:652]: send_publish(): NULL body Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua_mi [mi_func.c:290]: mi_pua_publish(): sending publish failed Kamailio verion is "version: kamailio 4.3.0 (x86_64/linux) c6aa95" I can issue other XMLRPC commands, urloc lookups method lists and those work, I just cant see where I am going wrong with this one. Any pointers would be greatly appreciated. Thanks

1 1

dmq/dmq_userloc crash on startup (kamailio 4.3.0)
by Asgaroth 10 Jul '15

10 Jul '15

Hi All, Another thing I noticed while playing with kamailio and dmq/dmq_usrloc is that 7 out of 10 times the kamailio process crashes when I initially start it up. Any tips/requests for further debugging would be greatly appreciated. Kamailio version: version: kamailio 4.3.0 (x86_64/linux) c6aa95 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c6aa95 compiled on 16:14:27 Jun 23 2015 with gcc 4.4.7 I start kamailio as follows, from commandline: /usr/sbin/kamailio -P /var/run/kamailio_registrar01/kamailio.pid -f /shared/kamailio/etc/kamailio_registrar.cfg -l "udp:(10.6.0.173, 10.7.0.173)" -m 1024 -M64 -A RESOURCE_REGISTRAR01 -l tcp:10.6.0.173:80 -u root -g root -ddd DMQ related settings are as follows: # grep -i dmq module_definitions.cfg loadmodule "dmq.so" loadmodule "dmq_usrloc.so" # ----- dmq params ----- modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:sip-registrars.internal_domain.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) # ----- dmq_usrloc params ----- modparam("dmq_usrloc", "enable", 1) DMQ_ADDRESS is defined as "sip:10.6.0.173:5060" in this servers case. sip-registrars.internal_domain.com resolves to the following address list 10.6.0.173, 10.6.0.174, 10.6.0.175 Here is the debug log of the crash: Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:82]: worker_loop(): dmq_worker [0 27445] getting lock Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:84]: worker_loop(): dmq_worker [0 27445] lock acquired Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=66fabe782d71b7a9842d4a6a5eb93c4d-1de2 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:454]: dmq_notification_callback(): dmq triggered from dmq_notification_callback Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.173:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.175:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.174:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:468]: dmq_notification_callback(): received 0 new or changed nodes Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 0 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 35 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 70 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:388]: usrloc_dmq_request_sync(): requesting sync from dmq peers Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:404]: usrloc_dmq_request_sync(): sending serialized data {"action":3} Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:226]: usrloc_dmq_send(): sending dmq broadcast... Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.6.0.173] == [10.6.0.173] Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:usrloc@10.6.0.174:5060> Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 24685 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.6.0.173] == [10.6.0.173] Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:usrloc@10.6.0.175:5060> Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 24686 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmq_funcs.c:163]: bcast_dmq_message(): skipping node sip:10.6.0.173:5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (10.6.0.174, 10.6.0.174, 0) Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:134]: worker_loop(): sent reply Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=66fabe782d71b7a9842d4a6a5eb93c4d-1ac2 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:250]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.6.0.174:5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:360]: usrloc_dmq_handle_msg(): Received DMQ_SYNC. Sending all contacts... And here is the backtrace from the core file: # gdb /usr/sbin/kamailio /core.27445 GNU gdb (GDB) Red Hat Enterprise Linux (7.2-75.el6) Copyright (C) 2010 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-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...(no debugging symbols found)...done. [New Thread 27445] Reading symbols from /lib64/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libdl.so.2 Reading symbols from /lib64/libresolv.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libresolv.so.2 Reading symbols from /lib64/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib64/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 /usr/lib64/kamailio/modules/db_unixodbc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/db_unixodbc.so Reading symbols from /usr/lib64/libodbc.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libodbc.so.2 Reading symbols from /usr/lib64/kamailio/libsrdb1.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libsrdb1.so.1 Reading symbols from /usr/lib64/kamailio/libkcore.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libkcore.so.1 Reading symbols from /usr/lib64/libltdl.so.7...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libltdl.so.7 Reading symbols from /lib64/libpthread.so.0...(no debugging symbols found)...done. [Thread debugging using libthread_db enabled] Loaded symbols for /lib64/libpthread.so.0 Reading symbols from /usr/lib64/kamailio/modules/mi_fifo.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/mi_fifo.so Reading symbols from /usr/lib64/kamailio/libkmi.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libkmi.so.1 Reading symbols from /usr/lib64/kamailio/modules/tm.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/tm.so Reading symbols from /usr/lib64/kamailio/modules/sl.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/sl.so Reading symbols from /usr/lib64/kamailio/modules/pv.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/pv.so Reading symbols from /usr/lib64/kamailio/libsrutils.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libsrutils.so.1 Reading symbols from /lib64/libm.so.6...(no debugging symbols found)...done. Loaded symbols for /lib64/libm.so.6 Reading symbols from /usr/lib64/kamailio/modules/kex.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/kex.so Reading symbols from /usr/lib64/kamailio/modules/usrloc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/usrloc.so Reading symbols from /usr/lib64/kamailio/modules/registrar.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/registrar.so Reading symbols from /usr/lib64/kamailio/modules/xlog.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/xlog.so Reading symbols from /usr/lib64/kamailio/modules/auth.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/auth.so Reading symbols from /usr/lib64/kamailio/modules/auth_db.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/auth_db.so Reading symbols from /usr/lib64/kamailio/modules/rr.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/rr.so Reading symbols from /usr/lib64/kamailio/modules/path.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/path.so Reading symbols from /usr/lib64/kamailio/modules/nathelper.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/nathelper.so Reading symbols from /usr/lib64/kamailio/modules/siputils.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/siputils.so Reading symbols from /usr/lib64/kamailio/modules/textops.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/textops.so Reading symbols from /usr/lib64/kamailio/modules/xmlrpc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/xmlrpc.so Reading symbols from /usr/lib64/libxml2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libxml2.so.2 Reading symbols from /lib64/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libz.so.1 Reading symbols from /usr/lib64/kamailio/modules/ctl.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/ctl.so Reading symbols from /usr/lib64/kamailio/modules/cfg_rpc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/cfg_rpc.so Reading symbols from /usr/lib64/kamailio/modules/dmq.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/dmq.so Reading symbols from /usr/lib64/kamailio/modules/dmq_usrloc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/dmq_usrloc.so Reading symbols from /lib64/libnss_files.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libnss_files.so.2 Reading symbols from /usr/lib64/gconv/ISO8859-1.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/gconv/ISO8859-1.so Reading symbols from /usr/lib64/psqlodbc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/psqlodbc.so Reading symbols from /usr/lib64/libssl.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libssl.so.10 Reading symbols from /usr/lib64/libpq.so.5...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libpq.so.5 Reading symbols from /usr/lib64/libodbcinst.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libodbcinst.so.2 Reading symbols from /lib64/libgssapi_krb5.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libgssapi_krb5.so.2 Reading symbols from /lib64/libkrb5.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5.so.3 Reading symbols from /lib64/libcom_err.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libcom_err.so.2 Reading symbols from /lib64/libk5crypto.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libk5crypto.so.3 Reading symbols from /usr/lib64/libcrypto.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libcrypto.so.10 Reading symbols from /lib64/libcrypt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libcrypt.so.1 Reading symbols from /lib64/libldap_r-2.4.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libldap_r-2.4.so.2 Reading symbols from /lib64/libkrb5support.so.0...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5support.so.0 Reading symbols from /lib64/libkeyutils.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libkeyutils.so.1 Reading symbols from /lib64/libfreebl3.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libfreebl3.so Reading symbols from /lib64/liblber-2.4.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/liblber-2.4.so.2 Reading symbols from /usr/lib64/libsasl2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libsasl2.so.2 Reading symbols from /usr/lib64/libssl3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libssl3.so Reading symbols from /usr/lib64/libsmime3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libsmime3.so Reading symbols from /usr/lib64/libnss3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libnss3.so Reading symbols from /usr/lib64/libnssutil3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libnssutil3.so Reading symbols from /lib64/libplds4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libplds4.so Reading symbols from /lib64/libplc4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libplc4.so Reading symbols from /lib64/libnspr4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libnspr4.so Reading symbols from /lib64/libselinux.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libselinux.so.1 Reading symbols from /lib64/librt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/librt.so.1 Reading symbols from /lib64/libnss_dns.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libnss_dns.so.2 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio_registrar01/kamailio.pid -f /shared/kam'. Program terminated with signal 11, Segmentation fault. #0 0x00007fdad84d76a5 in memcpy () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install kamailio-4.3.0-0.el6.bf.x86_64 (gdb) bt #0 0x00007fdad84d76a5 in memcpy () from /lib64/libc.so.6 #1 0x00007fdacfeff03d in usrloc_get_all_ucontact () from /usr/lib64/kamailio/modules/dmq_usrloc.so #2 0x00007fdacff02dfd in usrloc_dmq_handle_msg () from /usr/lib64/kamailio/modules/dmq_usrloc.so #3 0x00007fdad012d4a7 in worker_loop () from /usr/lib64/kamailio/modules/dmq.so #4 0x00007fdad011232c in ?? () from /usr/lib64/kamailio/modules/dmq.so #5 0x0000000000592675 in ?? () #6 0x00000000005923b0 in ?? () #7 0x000000000059297f in init_child () #8 0x00000000004a7680 in main_loop () #9 0x00000000004acd36 in main () (gdb) ^C(gdb) Quit

2 4

dialog: use keepalive & timeout features together
by Andrey Utkin 09 Jul '15

09 Jul '15

Would be nice if it would be possible to - make timer of "default_timeout" reset to 0 when keep-alive reply is received from remote part of call leg, so that the timeout expires (triggering call termination) only if some call party gets offline (we don't want just to limit the duration of normal call, even if call parties don't send any new SIP requests); OR - make unresponsiveness of call parties to keep-alive messages (controlled with ka_timer, ka_interval parameters) lead to some action, like call termination. Please advise if I miss something, and how to implement this better. Thanks in advance. -- Andrey Utkin

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Re: [SR-Users] Kamailio deployment
by Daniel Tryba 09 Jul '15

09 Jul '15

On Thursday 09 July 2015 17:46:22 Srujan Kumar K wrote: > I am connecting my kamailio server to MSC for my solution which has both > inbound and outbound. ... > Kamailio server IP : 172.22.14.12 > MSC IP : 172.28.0.68 Why does you kamailio config bind to the msc ipadress?

1 0

[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

11 24

Kamailio deployment
by Srujan Kumar K 09 Jul '15

09 Jul '15

Hai, I am connecting my kamailio server to MSC for my solution which has both inbound and outbound. The SIP trunk is not UP and we don't see any packets transmitted between the 2 nodes. We are also trying to load SIPI/SIPT module Kamailio server IP : 172.22.14.12 MSC IP : 172.28.0.68 I have also attached configuration file with this mail. Requesting your help on this. Regards ~Srujan

1 0

Moving from haproxy to db_cluster
by Jan Hazenberg 09 Jul '15

09 Jul '15

Hi, I have a question about the db_cluster module. We are currently using a galera mysql cluster from severalnines, in this setup haproxy is installed on the kamailio nodes handling the database failover. The question is if a database node fails and the TCP connection breaks the sql query will probably fail when using haproxy since it is not aware of the query. Will the db_cluster module send the query to the next available mysql node in the cluster without a failing query? If thats the case i should not notice the failing dbnode in my kamailio script. In other words, will there be a advantage in using the db_cluster module instead of haproxy? Thanks, Jan Hazenberg

2 3

code coverage in kamailio using gcov
by Surendra Pullaiah 09 Jul '15

09 Jul '15

Hello I am trying to implement code coverage tool in kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and it got compiled successfully. My problem is when I am running kamailio it is saying lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. Can anyone help me on this. Regards Surendra

1 0

Function sdp_remove_codecs_by_id seems to be not working
by José Seabra 09 Jul '15

09 Jul '15

Hello, I'm using the function sdp_remove_codecs_by_id from sdpops module in order to remove some codecs in INVITE request before send it to freeswitch, but the function doesn't remove the codec, and it doesn't give any error message. I'm using this function in request route. Kamailio version is 4.2.2. INVITE that kamailio receives from phone: INVITE sip:401@teste.d <sip%3A401(a)teste.itcenter.com.pt>emo.pt;user=phone SIP/2.0 Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0 Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060 From: "301" <sip:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt> >;tag=oztyflbzbx To: <sip:401@teste.demo.pt <sip%3A401(a)teste.itcenter.com.pt>;user=phone> Call-ID: 3c3a58a25d63-ghfc5xdg1sn0 CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1 X-Serialnumber: 000413262FA0 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:teste.demo.pt <http://teste.itcenter.com.pt> >;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 391 v=0 o=root 24935823 24935823 IN IP4 192.168.10.147 s=call c=IN IP4 192.168.10.147 t=0 0 m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv INVITE that kamailio send to freeswitch after execute sdp_remove_codecs_by_id("18"): INVITE sip:401@teste.demo.pt;user=phone SIP/2.0. Record-Route: <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>. Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Via: SIP/2.0/UDP 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0. Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0. Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060. From: "301" <sip:301@teste.demo.pt>;tag=zvjgcz9zs9. To: <sip:401@teste.demo.pt;user=phone>. Call-ID: 3c3a7c84e065-pr2hm0uk9yfz. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1. X-Serialnumber: 000413262FA0. P-Key-Flags: resolution="31x13", keys="4". User-Agent: snom370/8.4.35. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Call-Info: <sip:teste.itcenter.com.pt>;appearance-index=1. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 403. . v=0. o=root 228603317 228603317 IN IP4 100.64.250.4. s=call. c=IN IP4 100.64.250.4. t=0 0. m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:9 G722/8000. a=rtpmap:99 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:49405. SDP body has no changes related with codecs. Anyone call help please. Thank you BR José Seabra -- Cumprimentos José Seabra

3 6

Textops and Multipart Body - adding ISUP segfaults Kamailio
by andre second 09 Jul '15

09 Jul '15

Hi, I am using Kamailio version: kamailio 4.3.0 (x86_64/linux) c6aa95 on CentOS 6 I am trying to encapsulate ISUP in the INVITE: ...if(has_body("application/sdp")){ set_body_multipart(); if(msg_apply_changes()) { $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c"; append_body_part("$var(acm)","application/isup; version=itu-t92+","signal; handling=optional"); xlog("L_INFO", "ISUP Changes Applied Succesfully"); }} ...And getting the following error: Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1691]: get_boundary(): boundary is <--unique-boundary-1>...Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1808]: check_boundaries(): last bondary without -- at the end Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1612]: replace_body(): old size body[365] actual[534]Jul 7 14:59:12 ukrcld1000 kamailio: CRITICAL: <core> [data_lump.c:463]: free_lump(): non free-able lump: 0x7f7a33d5ae40 flags=1 and then Kamailio crashes. Can you please recommend? Thanks,Andrei

2 3

usleep with variable
by Antanas Masevicius 09 Jul '15

09 Jul '15

Hello, been struggling to make a random sleep with cfgutils functions. ex: $avp(i:444) = 123; usleep("$avp(i:444)"); gives an error: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio-sp1new.cfg:723 Kamailio version: 4.3 If usleep accepts only numbers, is there any other way to sleep by passing variable? best regards, Antanas -- # <#> Antanas Masevičius Technikos Direktorius UAB Nacionalinis telekomunikacijų tinklas Panerių g. 51, LT-03160 Vilnius *Tel.* 8 5 2056000 *Tiesiog. tel.* 8 700 00031 *Mob. tel.* 8 685 77730 *Faks.* 8 700 00034 *El. p.:* antanas.masevicius(a)ntt.lt <#> # www.ntt.lt <http://www.ntt.lt> | www.spykas.com <http://www.spykas.com>

1 1

MetaSwitch ignores Contact: header
by Markus 08 Jul '15

08 Jul '15

Dear list, I'm running kamailio-4.2.3-3.1.x86_64 in a very simple setup: one Asterisk server behind kamailio. But I'm using dispatcher, the other servers are just down currently. :) Different vendors route DIDs to kamailio's IP and kamailio handles the communication towards Asterisk. Now I have a new vendor who uses MetaSwitch, and there is a problem. All calls get dropped with a BYE from the Asterisk side after 32 seconds because Asterisk never receives the ACK, because kamailio doesn't forward it, because MetaSwitch sends it as ACK sip:number@KAMAILIO-IP instead of ACK sip:number@ASTERISK-IP That means MetaSwitch ignores what we tell it in the Contact: header. Is this MetaSwitch'es "fault" or kamailio's? All other vendors work fine without problems. Therefore I have compared an "old" vendor and the new one. Here is the result that shows the difference: Old vendor - works: OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 200 OK OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@ASTERISK-IP:5060 KAMAILIO-IP.5060 > ASTERISK-IP.5060: ACK sip:number@ASTERISK-IP:5060 New vendor with MetaSwitch - doesn't work: NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ... and so on ... Can I do something about this or is a change on MetaSwitch'es side in order? Thank you! Markus

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