- sr-users - kamailio.org

MySQL Database installation issue
by Kranti Kumar 15 Dec '15

15 Dec '15

Hi All, I am Newbie to Kamailio and while we configuring IMS Kamailio , facing issues on MySQL installation. OS : Debian 7 (Wheezy) While we issue apt-get install mysql-server-5.6, getting below error: apt-get install mysql-server-5.6 The following packages have unmet dependencies: mysql-community-server : Depends: libaio1 (>= 0.3.93) but it is not installable Depends: libnuma1 (>= 2.0.8~rc4) but it is not installable Recommends: mysql-client (= 5.6.28-1debian7) E: Unable to correct problems, you have held broken packages. Note:in mysql list added below lines: deb http://repo.mysql.com/apt/debian wheezy mysql-5.6 deb-src http://repo.mysql.com/apt/debian wheezy mysql-5.6 deb http://repo.mysql.com/apt/debian wheezy mysql-apt-config Can any one help us to resolve this problem. Thank you so much in advance. Regards, -kranti

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Re: [SR-Users] MySQL Database installation issue
by Kranti Kumar 15 Dec '15

15 Dec '15

Ganesh, After issuing command $sudo apt-get update, adding the repo we are getting the below error. root@debian:/etc/apt/sources.list.d# apt-get install mysql-server-5.6 Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'mysql-community-server' instead of 'mysql-server-5.6' Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: mysql-community-server : Depends: libaio1 (>= 0.3.93) but it is not installable Depends: libnuma1 (>= 2.0.8~rc4) but it is not installable Recommends: mysql-client (= 5.6.28-1debian7) E: Unable to correct problems, you have held broken packages. Regards, -kranti On Tue, Dec 15, 2015 at 8:20 AM, Infinicalls Infinicalls < infinicalls(a)gmail.com> wrote: > Hi, > Did you try issuing the command $sudo apt-get update after adding the > repo? What error message you are getting? > > Ganesh Kumar > > > On 12/15/15, Kranti Kumar <krantikumara(a)gmail.com> wrote: > > Hi All, > > > > I am Newbie to Kamailio and while we configuring IMS Kamailio , facing > > issues on MySQL installation. > > > > OS : Debian 7 (Wheezy) > > > > While we issue apt-get install mysql-server-5.6, getting below error: > > > > apt-get install mysql-server-5.6 > > > > > > The following packages have unmet dependencies: > > > > mysql-community-server : Depends: libaio1 (>= 0.3.93) but it is not > > installable > > > > Depends: libnuma1 (>= 2.0.8~rc4) but it is not > > installable > > > > Recommends: mysql-client (= 5.6.28-1debian7) > > > > E: Unable to correct problems, you have held broken packages. > > > > > > Note:in mysql list > > > > added below lines: > > > > deb http://repo.mysql.com/apt/debian wheezy mysql-5.6 > > > > deb-src http://repo.mysql.com/apt/debian wheezy mysql-5.6 > > > > deb http://repo.mysql.com/apt/debian wheezy mysql-apt-config > > > > > > Can any one help us to resolve this problem. > > > > Thank you so much in advance. > > > > Regards, > > > > -kranti > > > > > -- > > -- > http://www.infinicalls.com >

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Issue with BYE and $du
by Rene Montilva 14 Dec '15

14 Dec '15

Hi list What could be the scenario, when hangup a call and kamailio show empty the $du var and can't send BYE to endpoint -- Ing. Rene Montilva *FOSS Developer and VoIP Engineer.*

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how to send BYE to outbound leg in kamailio?
by Dileep Nunna 14 Dec '15

14 Dec '15

Hi, After credit limit reached(ims_charging), tried to send BYE to both legs in kamailo (modules/dialog_ng/dlg_req_within.c).Bye sent to caller, but when sending bye to callee it is failing.(it is considering record-routes from caller invite, not in 200 OK from callee).Is there any changes necessary? Can you please reply. Thanks, Dileep

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403 Not Relaying when default port changed
by Javohir Maxmarajabov 14 Dec '15

14 Dec '15

Hello fellow developers... We have been testing Kamailio for a week and it is working great... But some of the our friends reported that they can't connect to our server using their mobile Internet... and it seems default SIP port is blocked by Service Provider... So we decided to change ports... I opened kamailio.cfg and added these lines: listen=tcp:_PUBLIC_IP_:32850 listen=udp:_PUBLIC_IP_:32850 listen=tls:_PUBLIC_IP_:32851 listen=tcp:127.0.0.1:32850 listen=udp:127.0.0.1:32850 listen=tls:127.0.0.1:32851 listen=tcp:10.19.0.5:32850 listen=udp:10.19.0.5:32850 listen=tls:10.19.0.5:32851 port=32850 port=32851 Now clients throwing an error "403 Not relaying"... I tried to enable debugging to see cfgtrace... no errors detected But when I comment out lines above, (with default cfg) everything works fine Where and What am I doing wrong? Thanks in advance :) Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

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IMS-P-CSCF: stripped down version of kamailio.cfg
by Franz Edler 12 Dec '15

12 Dec '15

Dear IMS-experts, I am trying to reduce the complexity of the IMS implementation by disabling the functions which are not required for a pure stripped down IMS core platform. Therefore I deactivate NAT handling, RTP proxy, SBC, dispatcher etc... in a first step. Later on I will add the functions step-by step. According to the requirements. Now I discovered an inconsistency in the configuration file (kamilio.cfg) of P-CSCF. In main routing logic there is the point where P-CSCF distinguishes an originating-initial request from a terminating-initial request. This is done by the code: if (!ds_is_from_list()) { # Originating from Subscriber: route(Orig_Initial); } else { # Terminating to Subscriber: route(Term_Initial); } Unfortunately the function ds_is_from_list() is part of the dispatcher module which I want to avoid in a first step. Therefore another criterion must be used to distinguish between originating- and terminating-initial request. I therefore propose to use the pseudo-user-name "term" in the Route-URI as a distinguisher. What do you think? I am actually thinking about the correct statement (regular expression) to do that. Maybe anyone can help. BR Franz

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transformations {s.rm,match}
by Jonathan Hunter 11 Dec '15

11 Dec '15

Hi Guys, I have been using {s.rm,match} to remove occurrences of numbers, however I am aware it removes all occurrences. Is it possible to make it only remove the first occurrence and not all? For example If I am looking to modify 639157407639 I want to remove only the first 63, not both. So I end up with 9157407639 and not 91574079. Is it possible with a transformation of is regex the way ? thanks Jon

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Dispatcher Module & ds_load_list enhancement
by jay binks 11 Dec '15

11 Dec '15

Just an observation and enhancement suggestion. VERY occasionally Ive been getting *ERROR: <core> [parser/parse_param.c:591]: parse_params2(): parse_params(): No memory left* Now I know this is the root of the problem BUT ... it would be nice if it didnt leave my kamailio with no dispatchers at all. Im just suggesting that rather than calling destroy_list before re-populating the list. Could this be changed to create a new list, swap over the pointers to the new distributor list ( IF it has > 0 entries ) and then destroy_list . its just a thought ? -- Sincerely Jay

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Asynchronous processing increases branch count
by Alex Balashov 11 Dec '15

11 Dec '15

Hi, Normally with serial forking, the first branch ID corresponds to $T_branch_idx == 0, i.e. Via: SIP/2.0/UDP 172.30.110.4;branch=z9hG4bK7f3d.57593ee3a139f0ae6c4a8a0f531e342c.0 In the event of failure, a second branch attempt has a branch index of 1 and so on: Via: SIP/2.0/UDP 172.30.110.4;branch=z9hG4bK7f3d.57593ee3a139f0ae6c4a8a0f531e342c.1 However, when I use asynchronous processing with t_suspend() / t_continue(), resuming the transaction in an rtimer process, I notice that the $T_branch_idx of the first branch is 1, then 2. Where did 0 go? Is this a normal effect of TM suspend/continue? Thanks! -- Alex -- 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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problem with cfgtrace
by Joao Alves 11 Dec '15

11 Dec '15

Hi, Have just installed kamailio 4.3.4 and for some reason not getting cfgtraces from debugger. I've picked the default kamailio-advanced.cfg, and did no changes other than setting #!define WITH_DEBUG. Any suggestions on what could be causing this to happen? 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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problem with cfgtrace
by Joao Alves 10 Dec '15

10 Dec '15

Hi, Have just installed kamailio 4.3.4 and for some reason not getting cfgtraces from debugger. I've picked the default kamailio-advanced.cfg, and did no changes other than setting #!define WITH_DEBUG. Any suggestions on what could be causing this to happen? 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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WebRTC no longer supports RTP
by Yuriy Gorlichenko 10 Dec '15

10 Dec '15

https://developers.google.com/web/updates/2015/10/chrome-47-webrtc So at 47 chrome we already have no sound. What kind of proto we must use and how to handle this with rtpengine? Do anyone have same problems with it?

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info: kamailio.org updates - letsencrypt for https and dkim for email
by Daniel-Constantin Mierla 10 Dec '15

10 Dec '15

Hello, during the past few days I made some updates related to the security aspects of kamailio.org services. Two are relevant for the community. 1) First, kamailio.org uses now a TLS certificate signed by letsencrypt.org, a free trusted CA backed up by Mozilla and other internet companies, so browsing via HTTPS should no longer issue any warning of untrusted certificate (previously we used a CACert.org certificate which was not trusted automatically by browsers). Wiki and mailing lists portals use the letsencrypt certificate as well, so is no reason not to browse all kamailio.org and lists.sip-router.org pages only via HTTPS. Perhaps in the near future we will try to enable redirect of HTTP to HTTPS at least for the main page and login pages for wiki, mailing lists and other places that require sensitive data. Now SSLLabs test ranks https://kamailio.org with grade A: * https://www.ssllabs.com/ssltest/analyze.html?d=kamailio.org&latest As a side note, for those that haven't noticed it, for quite some time kamailio.org is available also over IPv6. 2) Second, emails forwarded by kamailio.org and lists.sip-router.org are having now a DKIM signature. Also, there are SPF records in DNS for these domains. Hopefully, those two will help getting the emails to be allowed by various spam filters out there, as their legit origin can be checked. If you check the sources of an email messages and the email server of receiving party is doing DKIM/SPF checks, you should see some headers like next (taken from an email I received to my gmail account from sr-users mailing list): """ Authentication-Results: mx.google.com; spf=pass (google.com: domain of sr-users-bounces(a)lists.sip-router.org designates 193.22.119.66 as permitted sender) smtp.mailfrom=sr-users-bounces(a)lists.sip-router.org; dkim=pass header.i=(a)lists.sip-router.org DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; c=relaxed/relaxed; d=lists.sip-router.org; s=20151206; h=Sender:Content-Type:List-Subscribe:List-Help:List-Post:List-Archive:List-Unsubscribe:List-Id:Reply-To:Subject:MIME-Version:Message-ID:To:From:Date; bh=lGjvCZYcxBHUHaJDnut1j2YTyPsXTnXHzUb0CgcDc1Q=; b=DlD+MKoEqyISB5Ba775t3zg70FC6ouC+tEo7j5zv4dn2Dhm4pWqkQXSfU4Kp1NqW1ZRYFC/mpg/7LEcGW2FlDL9J0FpUg1VjNmN7D1wvtW08hBBw91tsXImu9yf7KZjg/p4IbXu6vznldubrSxweIaV3q/xbrLgaqP5Dsrvs/9A=; """ Kamailio is not enforcing any of those policies on received email messages, so sending to the lists should not be affected. Should anyone discover problems when browsing the web portals or notices issues with emails from our mailing lists, report them to sr-dev mailing list. Also, if anyone has more hints on increasing the security/privacy for the web server and email systems we run for kamailio.org, do not hesitate to provide us suggestions. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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RTPPROXY unpredictable behavior
by Javohir Maxmarajabov 10 Dec '15

10 Dec '15

I am trying to enable RTPPROXY on debian, but it seems RTPProxy is ignoring my arguments... So, I am launching RTPPROXY using command: rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -p /var/run/rtpproxy.pid -R -a -P -r /tmp/rtppath -S /tmp/rtpspool -u rtpproxy rtpproxy Also tried with: rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -u rtpproxy rtpproxy And when I try to start Kamailio service I am getting error: 0(10562) ERROR: rtpproxy [rtpproxy.c:1681]: send_rtpp_command(): can't send command to a RTP proxy 0(10562) ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy does not respond, disable it 0(10562) WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP proxy 0(10562) WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy has been disabled temporarily p.s. I have checked RTPPROXY, and I am sure that it is launched ps aux | grep rtpproxy returned me: root 2770 0.0 0.0 19392 480 ? Ssl 04:59 0:00 /usr/bin/rtpproxy -l _MY_PUBLIC_IP_ -s udp:127.0.0.1 7722 -p /var/run/rtpproxy.pid -R -a -P -r /tmp/rtppath -S /tmp/rtpspool -u rtpproxy rtpproxy root 2775 0.0 0.0 11740 932 pts/1 S+ 04:59 0:00 grep --color=auto rtp Which says everything is ok and rtpproxy is running with arguments that I declared But the problem is, When I check it with: netstat -tulpn | grep rtpproxy I am receiving: udp 0 0 127.0.0.1:22222 0.0.0.0:* 2853/rtpproxy As you can see rtpproxy is listening on port 22222 and completely ignoring arguments I declared :( Where and What am I doing wrong? Sorry for my English and Thanks in advance... -- Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

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having a count of lost response for dialog keep alives
by Jayesh Nambiar 10 Dec '15

10 Dec '15

Hi, I'm using dlg_set_property(ka-src) and dlg_set_property(ka-dst) to keep alive my clients. In case of a network change the client does a Re-Invite with the new contact address and kamailio does keep-alives to the new address properly as the fix reported here: https://github.com/kamailio/kamailio/issues/273 The problem happens when the Re-Invite transaction on the client side is not complete and kamailio happens to send an OPTIONS keep-alive at that same time, the client fails to send back a 200 OK, because the client itself is trying to gain access on the new network. So as a work-around I thought would it be possible to end the dialog on missing at least 3 responses and not killing the dialog if one response is missed. Can it be a configurable parameter in the dialog module or something on those lines. Just a suggestion !! Thanks. - Jayesh

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random incorrect method parsing
by Vasiliy Ganchev 10 Dec '15

10 Dec '15

Hi folk! Have a strange issue, and cannot understand what is wrong. Test scheme UA(sip) -> INVITE -> Kamailio The transport protocol used is TCP. The issue is reproduced randomly, in case of wrong INVITE, Kamailio does not parse Method from R-URI and answer "400 CSeq method does not match request method". The log with debug level 3 for such INVITE: https://gist.github.com/vance-od/c4e1c783adba02d80c58 my first confuse is why in line 2 different number of bytes: "read= 1025 bytes, parsed=1037" the second is in line 12: Method value is empty after this everithing brokes and kamailio answers 400. Just another call: the same setup (same device, kamailio instance etc), same conditions, just another call. Everything is OK, similar part for good call: https://gist.github.com/vance-od/01b5dff2d81f0878cff4 All my attempts to find the diff between to messages goes to nothing, the 2 INVITEs looks similar, the wireshark analyse of TCP level - also had no result (all is similar) The issue happens only with one of our devices and only in case of TCP/TLS (UDP constantly ok, this is why I am assured the problem is in device, but I need proofs) version: kamailio 4.2.5 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, 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. will appreciate any help or idea ,with such a mess! Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/random-incorrect-method-parsing-tp1… Sent from the Users mailing list archive at Nabble.com.

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[Kamailio] Perfomance tests
by Prashant Desai 10 Dec '15

10 Dec '15

Hi, We are using Kamailio 4.2.3 and we have following query related to Performance tests. We see that kamailio wiki page has performance test results plotted for the Open SERV1.2.0. Would like to know is similar test are done for the later versions of kamailio till 4.2.3 ? If Yes please provide the link. Regards, Prashanth

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routing design
by Slava Bendersky 09 Dec '15

09 Dec '15

Hello Everyone, Looking for some design suggestion. I am trying achieve routing based on domain.tld. I thought use drouting, but not sure if it suitable for this task. Example: Kamailio Server 1 | |__________________ PBX Group 1 domain1.tld | |____ PBX Server 1 | |____PBX Server 2 |__________________ PBX Group 2 domain2.tld | |____ PBX Server 1 | |____PBX Server 2 and so on. Slava.

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Kamailio Redundancy Models
by Frank Costeira 08 Dec '15

08 Dec '15

Hi, I'm new to Kamailio and I'm trying to understand the architectural options and levels of redundancy that can be attained with Kamailio. In my particular scenario I would be using Kamailio as proxy server (no user registrations) to handle global prefix routing between 3 major regions; Americas, EMEA, and Aisa-Pacific. The idea would be to create a pool of Kamailio servers in each region. The pool would consist of two HA pairs. The HA pairs would be placed in two separate datacenter, and sessions would be distributed between the datacenter. In the event the Kamailio server failed it would have a local backup in the datacneter. In the event a datacenter failed, there would be an alternate datacenter available in the region. Can the LCR feature/module share a database, so that each kamailio server in the region have the same call routing information? Is there a feature or a way to automate the configuration of several Kamailio servers, when you want them to have the same routing logic/configuration. Is there a module, feature or API that would allow Kamailio to fetch a config, or possibly push the config via an API when the application is initialized. What is the best method for establishing a 1:1 back up with Kamailio? If we use TCP and route-via headers, the SIP session would be pinned through the active proxy server. Is there away to provide stateful failover so that the session state is maintained between primary and the backup server? Any pointers or recommendations would be highly appreciated. Regards, Frank

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secure websocket error in kamailio
by Priyaranjan Nayak 08 Dec '15

08 Dec '15

Hi All, I am running the kamailio for secure websocket (WSS). While running traffic testing the kamailio is not able to take beyond 2300 connection. I am getting below error. 45(12831) ERROR: tls [tls_server.c:243]: tls_fix_connection_unsafe(): tls: ssl bug #1491 workaround: not enough memory for safe operation: 208 45(12831) ERROR: sl [../../forward.h:247]: msg_send(): tcp_send failed 45(12831) ERROR: websocket [ws_handshake.c:113]: ws_send_reply(): sending reply 46(12832) ERROR: <core> [tcp_main.c:3577]: handle_ser_child(): received CON_ERROR for 0x7f03e0b0fa18 (id 651), refcnt 3, flags 0x4018 45(12831) WARNING: <core> [tcp_read.c:1642]: handle_io(): F_TCPCONN connection marked as bad: 0x7f03e0b0fa18 id 651 refcnt 1 44(12830) ERROR: tls [tls_server.c:154]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: 304 44(12830) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading I have seen below data in kamailio at the time of getting error. shmem:fragments = 57 shmem:free_size = 16986416784 shmem:max_used_size = 193649976 shmem:real_used_size = 193452400 shmem:total_size = 17179869184 shmem:used_size = 188516920 Could you please suggest me how can I get more performance from kamailio while using WSS ? Thanks Priyaranjan

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Proposing new logo for Kamailio project
by Daniel-Constantin Mierla 08 Dec '15

08 Dec '15

Hello, upon a process initiated before the summer during the IRC Devel Meeting, when we started to look for refreshing the logo of Kamailio project, we are now trying to finalize the decision and we are proposing a new logo for Kamailio project. The new proposed logo, as well as more details, can be seen at: * http://www.kamailio.org/w/2015/12/proposing-new-logo-for-kamailio-project/ We look forward to the feedback from community about this proposal - you like it or not, is too similar with other logos, etc... This announcement was sent to few of our mailing lists, but please reply to sr-users(a)lists.sip-router.org, being something concerning the community and definitely not something about development. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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lookup(location) offen dont work with websocket connected endpoints
by Yuriy Gorlichenko 08 Dec '15

08 Dec '15

long time ago looked this problem but resolve it by sql queries directly from confing file to location table. So while remember lookup(location) ofen can not find any contacts if its registers from websocket. it may work 10-15 times and then fails. So debug shows answer that is no connections in location table. Now at production system can not run this issue but at development system want to do it today or tomorrow.

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Installing 4.3 version from git
by Jurijs Ivolga 07 Dec '15

07 Dec '15

Hi all, I'm trying to install 4.3 version from git as described here: http://www.kamailio.org/wiki/install/4.3.x/git As far as I understood latest stable version is 4.3.3, but after installing from git I'm getting 4.3.4 version which is I believe is development version. kamailio -version version: kamailio 4.3.4 (x86_64/linux) 583dc0 Could you please check this? With kind regards, Jurijs

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crash on using http_query when response is 200
by Jayesh Nambiar 07 Dec '15

07 Dec '15

Hi, I'm using http_query to send a post request on a https URL. The weird part is it crashes with segmentation fault when the response from the server is 200. If the response was a 500, it works and continues processing. Here's the function: http_query("https://notify.abcd.com/onnet/call", "$avp(devices)", "Content-Type: application/json" "$var(response)"); xlog("L_INFO", "Response is $var(response)"); And here's the backtrace on crash: 0x00007f62f2bf8e38 in ?? () #1 0x00007f62ea22251b in http_query (_m=0x7f62f2c3a038, _url=0x7f62f2bf3548 "\330\064\277\362b\177", _dst=0x7f62f2bf9250 "\030\215\277\362b\177", _post=0x7f62f2bf35c8 "\200r\277\362b\177", _hdr=0x7f62f2bf91d0 "h\221\277\362b\177") at functions.c:226 #2 0x00007f62ea229e0b in w_http_query_post_hdr (_m=0x7f62f2c3a038, _url=0x7f62f2bf3548 "\330\064\277\362b\177", _post=0x7f62f2bf35c8 "\200r\277\362b\177", _hdr=0x7f62f2bf91d0 "h\221\277\362b\177", _result=0x7f62f2bf9250 "\030\215\277\362b\177") at utils.c:483 #3 0x000000000041f6b5 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2bf8a58, msg=0x7f62f2c3a038) at action.c:1079 #4 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2bf6fc0, msg=0x7f62f2c3a038) at action.c:1549 #5 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2bfb9b0, msg=0x7f62f2c3a038) at action.c:1045 #6 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2bb9388, msg=0x7f62f2c3a038) at action.c:1549 #7 0x000000000041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b8f248, msg=0x7f62f2c3a038) at action.c:678 #8 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b8e268, msg=0x7f62f2c3a038) at action.c:1549 #9 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b91040, msg=0x7f62f2c3a038) at action.c:1045 #10 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b8b7a8, msg=0x7f62f2c3a038) at action.c:1549 #11 0x000000000041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b3a2c0, msg=0x7f62f2c3a038) at action.c:678 #12 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b3a2c0, msg=0x7f62f2c3a038) at action.c:1549 #13 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b47f00, msg=0x7f62f2c3a038) at action.c:1045 #14 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b37e80, msg=0x7f62f2c3a038) at action.c:1549 #15 0x000000000041bc28 in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b1c9c8, msg=0x7f62f2c3a038) at action.c:678 #16 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b1c9c8, msg=0x7f62f2c3a038) at action.c:1549 #17 0x000000000041f3ce in do_action (h=0x7ffeb128c8c0, a=0x7f62f2b1cc28, msg=0x7f62f2c3a038) at action.c:1045 #18 0x000000000042c2ea in run_actions (h=0x7ffeb128c8c0, a=0x7f62f2b170a0, msg=0x7f62f2c3a038) at action.c:1549 #19 0x000000000042ca93 in run_top_route (a=0x7f62f2b170a0, msg=0x7f62f2c3a038, c=0x0) at action.c:1635 #20 0x000000000051c115 in receive_msg ( buf=0xab7520 <buf> "INVITE sip:bdbbf600000001b5b919d0@devsip.abcd.com:8321 SIP/2.0\r\nRecord-Route: <sip:203.153.53.145:8300;r2=on;lr=on>\r\nRecord-Route: <sip:203.153.53.145:8321;transport=tcp;r2=on;lr=on>\r\nVia: SIP/2.0/UDP"..., len=1901, rcv_info=0x7ffeb128cc10) at receive.c:196 #21 0x000000000062d427 in udp_rcv_loop () at udp_server.c:495 #22 0x00000000004b28c1 in main_loop () at main.c:1593 #23 0x00000000004b9aa5 in main (argc=5, argv=0x7ffeb128d118) at main.c:2597 Can someone please confirm the crash so that I'll go ahead and report this as an issue on the tracker. Basically when it crashes, kamailio doesn't even execute the xlog function after the http_query call. P.S. abcd.com is a changed domain for confidential reasons. Thanks, - Jayesh

3 2

PDB module long prefixes
by Igor Potjevlesch 07 Dec '15

07 Dec '15

Hi, I had configured my pdb module to accept 5 number prefixes by doing: In pdbt.c: bufsize = slen + 1 + 1 + *4* + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) ... ret = snprintf(p,* 7*, "%d\n", node->carrier); if (ret < 1 || ret > *6*) { And in common.h: #define MAX_PDB_CARRIERID *99999* And i hadn't had any problem until now. I had rows in my .csv like: 0645052050;10122 And it worked. The problem I have now is with prefixes bigger than 40000. I get segmentation fault when doiong a query with pdbt, however the pdbt build works good without any error. At first i thought of a buffer problem but eventhough testing with a .csv of 2 rows with 50000 prefixes will not work. Another curious effect is that for a prefix like 84238 the server changes the answer to 18754. I dont know why... Could anyone help me please? Thanks and regards, Igor.

1 0

Executing a C function from the Kamailio Script
by Mack Hendricks 07 Dec '15

07 Dec '15

We have some logic in a C function that we would like to execute from the Kamailio script. I see there are modules for calling functions in other languages. But, I don’t see anything for C. Is creating a module the only way to leverage this C function? Thanks, -Mack

4 4

SIP Trunking Kamailio directly to external SIP Trunk Provider
by amjadali_bb＠hotmail.com 07 Dec '15

07 Dec '15

Hi, I have got two Asterisk servers and both are communicating with the SIP Trunk provider (External Network) directly - the Asterisk servers are connected with the Kamailio and then Kamailio with the Agent Phones. I need some change in this architecture that I want Kamailio to communicate with the SIP Trunk provider (External Network) directly. Can any one please advise if this is possible and how ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/SIP-Trunking-Kamailio-directly-to-e… Sent from the Users mailing list archive at Nabble.com.

2 1

Announcement: Next Kamailio World - May 18-20, 2016, in Berlin
by Daniel-Constantin Mierla 07 Dec '15

07 Dec '15

Hello, a short message to announces that the date and location for next Kamailio World Conference were decided - May 18-20, 2016, in Berlin, Germany. Kamailio project is celebrating 15 years of development in 2016, therefore we plan to have a special edition. Website of the event and call for presentations will be launched very soon. Meanwhile, if you haven't participated at a past edition, you can check the previous edition website in order to get an idea about the structure and content of the event: - http://www.kamailioworld.com/k03/ Have a great week, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Kamailio radius authentication prolem
by Volkan Oransoy 07 Dec '15

07 Dec '15

Hi all, I try to authenticate my users via mod_radius, but I have problem. FreeRadius server gives this error: Auth: [digest] Cleartext-Password or Digest-HA1 is required for authentication. I think I need to send those attributes from kamailio but I couldn't figure out how to do it. Here is diff of my config with default config. Thanks, /Volkan ===================== diff /etc/kamailio/kamailio.cfg /etc/kamailio/kamailio.cfg.original < #!define WITH_DEBUG 294,297d292 < loadmodule "auth_radius.so" < modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") < loadmodule "avpops.so" < 739,783c734,739 < if (is_method("REGISTER")) < { < avp_print(); < if (!radius_www_authorize("example.com")) { < xlog("SCRIPT: www auth return code: $rc\n"); < switch ($rc) { < case -7: < send_reply("500", "Server Internal Error"); < exit; < case -1: < send_reply("400", "Bad Request"); < exit; < default: < }; < if (defined($avp(digest_challenge)) && < ($avp(digest_challenge) != "")) { < append_to_reply("$avp(digest_challenge)"); < }; < send_reply("401", "Unauthorized"); < exit; < }; < } < < if (from_uri==myself) < { < if (!radius_proxy_authorize("example.com", "$pU")) { # Realm and URI user are taken < switch ($rc) { # from P-Preferred-Identity < case -7: # header field < send_reply("500", "Server Internal Error"); < exit; < case -1: < send_reply("400", "Bad Request"); < exit; < default: < }; < if (defined($avp(digest_challenge)) && < ($avp(digest_challenge) != "")) { < append_to_reply("$avp(digest_challenge)"); < }; < send_reply("407", "Proxy Authentication Required"); < exit; < }; < < } < --- > #!ifdef WITH_IPAUTH > if((!is_method("REGISTER")) && allow_source_address()) { > # source IP allowed > return; > } > #!endif 784a741,753 > if (is_method("REGISTER") || from_uri==myself) > { > # authenticate requests > if (!auth_check("$fd", "subscriber", "1")) { > auth_challenge("$fd", "0"); > exit; > } > # user authenticated - remove auth header > if(!is_method("REGISTER|PUBLISH")) > consume_credentials(); > } > # if caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here

2 1

No NOTIFY while PUBLISHing new presence status
by Thierry Luo 07 Dec '15

07 Dec '15

Hi All, I'm now using Kamailio as Presence Server for my IM system. All is fine except that, when one user publishes new presence status (in tuples) to the Presence Server, which does not NOTIFY the subscribers of the status. And the more weird is that, when I restart the subscriber UAs and re-subscribe, the Presence Server send out the NOTIFY messages to the subscribers that just re-subscribed. This is not compliant with RFCs. Anybody please help me out. Attached is the configuration file. Thanks. Thierry Luo

2 2

Table 'kamailio.version' doesn't exist
by Annus Fictus 07 Dec '15

07 Dec '15

Hello List, when I try to create Kamailio Database, I receive this error: *kamdbctl create* MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... ERROR 1146 (42S02) at line 1: Table 'kamailio.version' doesn't exist ERROR: Creating core tables failed at standard! Kamailio version 4.3.4 Any hint? Regards

2 3

UAC dynamic reg_contact_addr()
by Yuriy Gorlichenko 07 Dec '15

07 Dec '15

I have multiple ip addresses at my kamailio. I use uac module for registration to sip providers. I have one provider but want to register form different ip addresses used by my server. When register sends it takes ip address form reg_contact_addr(). But if I want to register from another interface with different ip i need to get to this parametr another ip address. So As I understand it imposibleto do with avp. Does kamailio have some methods to do this?

2 3

why is rtpenengine-delete deleting the whole call?
by Juha Heinanen 05 Dec '15

05 Dec '15

in my test sip proxy parallel forks the invite to two contacts and two rtpengine-offers are made with these kind of call-id and via-branch params: "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0" "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1", then the via-branch .0 callee answers and rtpengine-answer is sent to with this kind of call-id and via-branch params: "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0", after that sip proxy deletes the other branch by sending rtpengine-delete with this kind of params: Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Received command 'delete' from 127.0.0.1:40700 Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Dump for 'delete' from 127.0.0.1:40700: { "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1", "received-from": [ "IP4", "127.0.0.1" ], "from-tag": "3322319973f3909e", "command": "delete" } and rtpengine responds: Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Response dump for 'delete' to 127.0.0.1:40700: { "created": 1448583951, "last signal": 1448583955, "tags": { "3322319973f3909e": { "tag": "3322319973f3909e", "created": 1448583951, "in dialogue with": "0f6759297b8f1711", "medias": [ { "index": 1, "type": "audio", "protocol": "RTP/AVP", "streams": [ { "local port": 53864, "endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10458 }, "advertised endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10458 }, "last packet ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... ": 1448583951, "flags": [ "RTP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } }, { "local port": 53865, "endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10459 }, "advertised endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10459 }, "last packet": 1448583951, "flags": [ "RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } } ], "flags": [ "initialized", "send", "recv" ] } ] }, "0f6759297b8f1711": { "tag": "0f6759297b8f1711", ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0", "created": 1448583951, "in dialogue with": "3322319973f3909e", "medias": [ { "index": 1, "type": "audio", "protocol": "RTP/AVP", "streams": [ { "local port": 53844, "endpoint": { "family": "IPv4", "address": "10.0.2.15", "port": 10144 }, "advertised endpoint": { "family": "IPv4", "address": "10.0.2.15", "port": 10144 }, "last packet": 1448583951, "flags": [ "RTP", "RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... } }, { "local port": 53845, "endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "advertised endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "last packet": 1448583951, "flags": [ "RTCP", "fallback RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } } ], "flags": [ "initialized", "send", "recv", "rtcp-mux" ] } ] } }, "totals": { "RTP": { "packets": 0, "bytes": 0, "errors": 0 }, "RTCP": { "packets": 0, "bytes": 0, "errors": 0 } }, "result": "ok" } which then results in deletion of the whole call, i.e., also the branch that answered. question: why does rtpengine not do what it is asked to do in rtpengine-delete, i,e. delete the offer corresponding to "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1"? -- juha

3 11

Kamailio + Asterisk, same machine, same public IP.
by Douglas Adami 04 Dec '15

04 Dec '15

Hello everyone, I have Kamailio and Asterisk on the same machine with the same public IP. When I try to forward calls to the asterisk (PSTN GW on port 5080), returns the error 401, but does not reach the Asterisk any information. When I switch to an IP out, it works. Any idea? Regards, Doug

2 6

If we print out $fu, $fU & $fd on onsend_route, it prints out incorrect values.
by Jurijs Ivolga 04 Dec '15

04 Dec '15

Hi all, I opened a bug here: https://github.com/kamailio/kamailio/issues/430 But then I was advised to write to mailing list. In nutshell: I'm running Kamailio 4.3.3. When I'm trying to update $ru, $fu & $fd in Kamailio conf: $rU = "1000"; $fu = "sip:something@mydomain.com"; $fd = "newdomain.com"; And later I'm trying to print out $rU, $fu, $fU & $fd in onsend_route: xlog("ALERT: From URI $fu \n"); xlog("ALERT: From domain $fd \n"); xlog("ALERT: From user $fU \n"); xlog("ALERT: Request user $rU \n"); Kamailio 4.3.3 prints out following: ALERT: From URI sip:2000@some_ip ALERT: From domain some_ip ALERT: From user 2000 ALERT: Request user 1000 But I believe it should: ALERT: From URI sip:something@mydomain.com ALERT: From domain mydomain.com ALERT: From user something ALERT: Request user 1000 I'm using default Kamailio config with minor changes, you can check it in attachment. Is it proper behavior? Thank you! With kind regards, Jurijs

2 1

[permissions] allow_source_address_group() always return -1
by Iñaki Baz Castillo 04 Dec '15

04 Dec '15

Hi all, Kamailio 4.3 (Debian Wheezy package) with db_mysql and permissions modules enabled. Once started, this is the cached content of the "address" table: ~# kamctl mi "address_dump" 22 <20, 99.23.86.78, 0> [TEST SERVER] Within the script: ----------------------------- xlog("L_INFO", "request_route() [method:$rm, si:$si, sp:$sp]\n"); $var(group) = allow_source_address_group(); if ($var(group) != -1) { // ALWAYS HERE :( xlog("L_WARN", "invalid source IP [si:$si]\n"); } ----------------------------- Then I send an INVITE from such an allowed IP, and got: INFO: <script>: request_route() [method:INVITE, si:99.23.86.78, sp:7060] WARNING: <script>: invalid source IP [si:99.23.86.78] "version" table contains: table_name table_version ------------------------------------- address 6 trusted 5 I've used code similar to the above in ~30 kamailios (however never in 4.X). Do I miss something? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 2

Re: [SR-Users] About Siremis Login Page Problem
by Daniel-Constantin Mierla 04 Dec '15

04 Dec '15

Hello, do you have the mod_redirect enabled for apache? Can you look at the logs of apache and see if there is any error message? Also, you can look at the logs of siremis in the folder siremis/logs/. Cheers, Daniel On 03/12/15 12:49, Morshed wrote: > > Dear Concern, > > I have installed kamailo and siremis installed properly. But I cannot > login to the siremis, the login page is not loaded successfully. > Please have a look on the attached image and any suggestion from your end? > > > > Regards, > > > > Morshed Alam > > Support Engineer | IP Telephony > > > > Link3 Technologies Ltd. > > Bulu Ocean Tower, 17th Floor > > 40, Kemal Ataturk Avenue, Banani, > > Dhaka 1213, Bangladesh. > > Tel: +88-02-9822288 > > IP Phone : +88-09678997705 > > Fax: +88-02-9821332 > > Web: www.link3.net <http://www.link3.net> > > > > > > _______________________________________________ > 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 http://miconda.eu

1 0

LCR module : re load_gws uri errors
by Yasin CANER 04 Dec '15

04 Dec '15

3 7

Generating db-tables .html file
by smititelu 04 Dec '15

04 Dec '15

Hello, I want to add a new table to the [1]. I managed to create a new *kamailio-*.xml* in "lib/srdb1/schema/", run make and see the resulting **-dbschema.xml* in doc/databases/kamailio, along with the other .sql scripts. Here I got stuck; I can't find the scrip that generates the final .html. I tried grepping the code for some keywords found in page source without success. Also, couldn't find the answer reading [2]. Any ideas how the final .html is generated? Thanks, Stefan [1] http://kamailio.org/docs/db-tables/kamailio-db-devel.html [2] http://www.kamailio.org/dokuwiki/doku.php/development:db-schema

2 6

terminate call when dialog when remote contact disappears without BYE
by Thibault Gueslin 03 Dec '15

03 Dec '15

I would like to terminate the dialog when one party disconnects without sending a BYE (no more connctivity, client shutdown) I am using TCP connection. Should I terminate the dialog from event_route[usrloc:contact-expired] ? I have tried also to use SIP OPTION parameter from dialog module modparam("dialog", "ka_timer", 10 ) modparam("dialog", "ka_interval", 30 ) but nothing is sent) thibault

3 13

SQLOPS vs APP_LUA
by Luke Milbourne 03 Dec '15

03 Dec '15

Hi All, I need to run some SQL and then set the destination URI according to the results, if any. I've got this working using APP_LUA, however, I've noticed SQLOPS. Can anyone advise on which method is likely to produce better results. I'm assuming my Lua function is connecting to the DB and then disconnecting each time. If I use SQLOPS, does this mean Kamailio will keep a connection available to the DB? -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

2 2

About Siremis Login Page Problem
by Morshed 03 Dec '15

03 Dec '15

Dear Concern, I have installed kamailo and siremis installed properly. But I cannot login to the siremis, the login page is not loaded successfully. Please have a look on the attached image and any suggestion from your end? Regards, Morshed Alam Support Engineer | IP Telephony Link3 Technologies Ltd. Bulu Ocean Tower, 17th Floor 40, Kemal Ataturk Avenue, Banani, Dhaka 1213, Bangladesh. Tel: +88-02-9822288 IP Phone : +88-09678997705 Fax: +88-02-9821332 Web: <http://www.link3.net> www.link3.net

1 0

using APP_LUA for complex arithmetic operations
by Jayesh Nambiar 03 Dec '15

03 Dec '15

Hi, I understand that kamailio does not allow float-point values in variables and I think kamailio is not a great place to do complex arithemtic operations and so I thought of using app_lua for the same. So can I pass the control to app_lua and come back to config script with some values? For eg. can I call a lua script that does a complex arithemetic operation on my AVP values and based on certain conditions; return back a true or false and then I continue in my existing script. I came across an exmaple http://kb.asipto.com/kamailio:usage:k32-lua-routing but this is more like doing the entire thing inside LUA. What I'm looking at is getting certain things done in lua and continue the script processing. Thanks for any help here. - Jayesh

3 4

Kamailio - Asterisk, problem with SIP trunk mode
by Mickael MONSIEUR 02 Dec '15

02 Dec '15

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

3 6

PKG memory leak in 4.2.6
by Julia Boudniatsky 02 Dec '15

02 Dec '15

Hello, We are running Kamailio 4.2.6 ( db77ac) compiled with memory debugging. The free PKG memory has decreased on ~6.5 M after two weeks. Please see attached file with the log messages at kamailio shutdown. Thank you in advance, Julia.

2 5

Record-route on sequential requests
by Kristian F. Høgh 01 Dec '15

01 Dec '15

Hi list, Record-route on sequential requests doesn't have did/vsf parm. On initial INVITE, I call record_route() and set dialog flag. kamailio adds the following RR Record-Route: <sip:ww.xx.yy.zz;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2> UAC sends a re-INVITE, containing the following route: Route: <sip:178.21.251.54;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2> I call loose_route() followed by record-route(), and kamailio adds the following RR: Record-Route: <sip:ww.xx.yy.zz;lr;ftag=15af612df> I've got a phone which updates the routeset on sequential requests (which I don't think it should) and the did-matching fails. Should kamailio add did/vsf on sequential requests, when I call record_route()? Should I set dlg_match_mode to 1? (I use 0) Is it wrong to call record_route() from sequential requests? Which way is the best to resolve the problem? Regards, Kristian Høgh Uni-tel A/S

2 4

Problem with NATed UAC
by Vladimir Stepic 01 Dec '15

01 Dec '15

Hello everyone, this is my first message. I'm glad to become a part of this big community. I'm new in SER technology. I worked in Asterisk PBX only, until now. For the first time i installed Elastix MT witch have kamailio in front of asterisk, and now i have a problem. I attach a diagram of initial configuration in my lab and pcap file and kamailio.cfg. Regardless diagram, x.x.x.x addresses are public and y.y.y.y are private. When i test connection 1.1 is working, connection 1.2 not working connection 2.2 not working connection 1.3 not working connection 2.3 working Phones are connected(authenticate) to kamailio, but when i look at asterisk cli those phones are unreachable, but can call, but can not be called. On router where android works, Siemens dont, and where pc softphone works other UAC dont work. I think that my nat-traversal configure good, regardless of documentation. Please tell me, if someone has some ideas of solve this problem. Thanks in advance, and sorry for my bad english :) ---------------------------------------------------------------------- *Vladimir Stepić,* Engineer specialist in computer networking System Linux Administrator mob. tel. +381 69/18-68-431 email: e1.e398(a)gmail.com

2 1

kamailio's Makefile system : Enabling CFLAGS/LDFLAGS for udp_server.c
by aft 01 Dec '15

01 Dec '15

Hi, I'm doing some experimentation on kamailio's udp_server.c. Now i need libpcre for that. So i first thought if i enable the regex module, the CFLAGS/LDFLAGS would be available from udp_server.c. But it turns out it doesn't build(doesn't *link* to be more precise). Kamailio's build system is pretty unfamiliar for me. Although i absolutely hate autotools based build system(I prefer Scons), but i'm pretty familiar with these. But Kamailio's build system is pretty much cryptic for me(Is it based on CMake?). There is a page on devel wiki for kamailio's build system. it turns out i can do stuff like : make cfg prefix=/opt/kamailio \ SCTP=1 \ CC_EXTRA_OPTS=-I/usr/gnu/include \ group_include="standard postgres presence" \ include_modules="snmpstats lcr tls" So there might be a LD_EXTRA_OPTS or stuff like that. Which than would be defined (I guess) globally so the core c files(like udp_server.c) would be able to use that. One way i guess would be to sharpen my egrep and find that variable name. But it all feels a bit hacky. As i have already included regex module, there should be a way to reuse the CFLAGS/LDFLAGS for that module for core files like udp_server.c. Cheers Arif

2 1

Re: [SR-Users] Nathelper module
by Daniel-Constantin Mierla 01 Dec '15

01 Dec '15

Hello, On 30/11/15 14:52, Marino Mileti wrote: > > I would like to use nathelper module in order to "ping" all registered > client and "delete" (from locations table) clients that are not > reachable (maybe there's some network issue, or some other problem) > > > > I've seen that it's possible to have this feature for natted contact > using nathelper (but i haven't tried this functionaliy) > > > > Is there a way to remove records from location table without wait that > its expire is 0? > > see: http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.p.… Also, look at the parameter of usrloc module for handling contacts that have the tcp connection cut. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Help Required
by anakweze charles 30 Nov '15

30 Nov '15

Hi All. Does anyone know where to find training guide for Kamailio HA? I want to have 2 Kamailio in HA with four (4) Freeswitch servers behind them. I will also like to customer cgrate to manage the accounting side in a multi-tenant environment. I am just researching for this on my own and I will appreciate a guide to realize this objective. Thanks

1 0

SNMPstats dialog 0
by Igor Potjevlesch 30 Nov '15

30 Nov '15

Hello, I have rebooted my kamailio machine and now the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 And also the status is: KAMAILIO-MIB::kamailioDialogUsageState.0 = INTEGER: idle(0) How can I change it to active(1)? I have dialog.so loaded as well as modparam("dialog", "dlg_flag", 4) and dlg_manage(). Thank you. Regards,

3 5

Reverted Error Message
by Sam Ware 30 Nov '15

30 Nov '15

Daniel, I got the following error message (currently running version 4.3): Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR: dispatcher [dispatcher.c:517]: mod_init(): call load dispatching DST ID_AVP set but no size for hash table Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing modul e dispatcher (/usr/local/lib64/kamailio/modules/dispatcher.so) I noticed in the following DEV GIT (Version 3.1) message that the error message was changed to a message that more clearly defined the error and correction for it. But the my current message would indicate that it got changed back to the older error message. http://lists.sip-router.org/pipermail/sr-dev/2010-November/009799.html Just thought I would point it out because the newer message from the GIT message seems clearer to me. -- *Sam D Ware* CIO *Vera Networks LLC* Email: sam.ware(a)veranetworks.com

2 1

Seremis Manual
by Omnes Leroy 30 Nov '15

30 Nov '15

Hi, Is there a manual I can get so Seremis. I am a new user. Thanks, Omnes

2 1

Kamailio geolocation module
by Mititelu Stefan 28 Nov '15

28 Nov '15

Hi everyone, Is there such a thing like "geolocation" module in Kamailio? (or something related to it i.e. [1] [2]). Thanks, Stefan Mititelu [1] https://tools.ietf.org/html/rfc6442 [2] https://tools.ietf.org/html/rfc3693#page-6

2 4

crash on using the $expires variable
by Jayesh Nambiar 28 Nov '15

28 Nov '15

Hi, I am trying to extract out the expires value from the incoming REGISTER request. It might be present in the Contact header or as a separate Expires header and hence I planned to use the $expires(max) PV to get this. But kamailio just crashes as soon as I use the variable. Here's when it crashes: if(is_method("REGISTER")) { xlog("L_INFO", "Expires is $expires(max)"); if(!save("location", "0x01")) { t_reply("500", "Error while saving AOR"); } } It crashes at the xlog point. Here's the backtrace for reference: Reading symbols from /usr/local/kamailio/sbin/kamailio...done. [New LWP 16609] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid -f /usr/local/kamail'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f34b658b311 in pv_get_expires (msg=0x7f34bd879e58, param=0x7f34bd7e9220, res=0x7ffcf67af530) at pv_core.c:3060 3060 if(c->expires && c->expires->body.len) { (gdb) backtrace #0 0x00007f34b658b311 in pv_get_expires (msg=0x7f34bd879e58, param=0x7f34bd7e9220, res=0x7ffcf67af530) at pv_core.c:3060 #1 0x00000000004fb6e1 in pv_get_spec_value (msg=0x7f34bd879e58, sp=0x7f34bd7e9208, value=0x7ffcf67af530) at pvapi.c:1304 #2 0x00000000004fbb46 in pv_printf (msg=0x7f34bd879e58, list=0x7f34bd80ef38, buf=0x7f34bd8735a0 "Expires is ", len=0x7ffcf67af618) at pvapi.c:1363 #3 0x00007f34b67ea745 in xl_print_log (msg=0x7f34bd879e58, list=0x7f34bd80ef38, buf=0x7f34bd8735a0 "Expires is ", len=0x7ffcf67af618) at xl_lib.c:37 #4 0x00007f34b67eb0f9 in xlog_helper (msg=0x7f34bd879e58, xm=0x7f34bd808648, level=2, line=0, facility=-1) at xlog.c:215 #5 0x00007f34b67ec160 in xlog_2_helper (msg=0x7f34bd879e58, lev=0x7f34bd7e9130 "", frm=0x7f34bd808648 "84\177", mode=0, facility=-1) at xlog.c:284 #6 0x00007f34b67ec19c in xlog_2 (msg=0x7f34bd879e58, lev=0x7f34bd7e9130 "", frm=0x7f34bd808648 "84\177") at xlog.c:292 #7 0x000000000041e23b in do_action (h=0x7ffcf67b0b40, a=0x7f34bd817cd0, msg=0x7f34bd879e58) at action.c:1065 #8 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd813e18, msg=0x7f34bd879e58) at action.c:1548 #9 0x000000000041ab16 in do_action (h=0x7ffcf67b0b40, a=0x7f34bd7e8738, msg=0x7f34bd879e58) at action.c:677 #10 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd7e8738, msg=0x7f34bd879e58) at action.c:1548 #11 0x000000000041e094 in do_action (h=0x7ffcf67b0b40, a=0x7f34bd7e89c8, msg=0x7f34bd879e58) at action.c:1044 #12 0x000000000042a8e7 in run_actions (h=0x7ffcf67b0b40, a=0x7f34bd7e19d8, msg=0x7f34bd879e58) at action.c:1548 #13 0x000000000042b04d in run_top_route (a=0x7f34bd7e19d8, msg=0x7f34bd879e58, c=0x0) at action.c:1634 #14 0x000000000050f3d4 in receive_msg ( buf=0xa80c00 <buf> "REGISTER sip:devsip.tlkn.com SIP/2.0\r\nVia: SIP/2.0/UDP 203.153.53.145:8300;branch=z9hG4bK119a.e23ed10b856f38b41c93d1d259d8869f.0;i=b1\r\nVia: SIP/2.0/TCP 172.16.16.32:52358;rport=52359;received=203.153."..., len=712, rcv_info=0x7ffcf67b0e60) at receive.c:196 #15 0x0000000000613b9e in udp_rcv_loop () at udp_server.c:495 #16 0x00000000004aa119 in main_loop () at main.c:1577 #17 0x00000000004b0e1e in main (argc=5, argv=0x7ffcf67b1318) at main.c:2553 Thanks, - Jayesh

3 4

restore_uri_reply() called twice in 200OK to BYE
by Efelin Novak 27 Nov '15

27 Nov '15

Hi Folks, I have a problem with To and From headers in 200 OK to BYE request. Their URI is doubled when leaving Kamailio. Incoming 200 OK: From : <sip:00421111@IP_ADDRESS>;tag=as017a2986 To : <sip:00421222@IP_ADDRESS0>;tag=K7ryDgH2g7gFm Outgoing 200 OK: From : <sip:+421111@IP_ADDRESS><sip:+421111@IP_ADDRESS>;tag=as017a2986 To : <sip:+421222@IP_ADDRESS><sip:+421222@IP_ADDRESS>;tag=K7ryDgH2g7gFm I'm using uac module to replace these headers in auto and dialog mode: modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) append_fromtag is also set: modparam("rr", "append_fromtag", 1) When I run kamailio in debug mode I can see the function restore_uri_reply() is called twice: . . . 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 234, <received> = <IP_ADDRESS>; state=6 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK0bb8c5b2>; state=6 0(3133) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <5060>; state=16 0(3133) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 0(3133) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=62 0(3133) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=teHD72X67aQUm 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 0(3133) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [60]; uri=[sip:421222@IP_ADDRESS:5060] 0(3133) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:421222@IP_ADDRESS:5060>] 0(3133) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <103> <BYE> 0(3133) DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 10806 label 0 branch 0 0(3133) DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0x220071b4)! 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x220071b4, callback type 2, id 0 entered 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=as46bccb83 0(3133) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421222@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421222@IP_ADDRESS>> 0(3133) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x220071b4, callback type 2, id 0 entered 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421111@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:705]: restore_uri_reply(): removing <<sip:00421222@IP_ADDRESS>> 0(3133) DEBUG: uac [replace.c:714]: restore_uri_reply(): inserting <<sip:+421222@IP_ADDRESS>> . . . I think the problem is in registering a function restore_uris_reply() in callbacks in functions (both in replace.c of uac module) * void rr_checker(struct sip_msg *msg, str *r_param, void *cb_param) * static void replace_callback(struct dlg_cell *dlg, int type, struct dlg_cb_params *_params) Both of these callbacks are registered for BYE and ACK requests (INVITE request is using different callback), therefore running restore_uri_reply() twice. For testing purposes after I have removed the callback in rr_checker and recompile the uac module, To and From headers were OK. In my configuration I'm calling uac_replace_to in RELAY route: route[RELAY] { if ( !has_totag() ) { dlg_manage(); uac_replace_to("$ru"); } if (!t_relay()) { sl_reply_error(); } } So I think there is a bug in uac module, when one function is registered twice in callbacks for NON-INVITE requests. Or is there a problem with my configuration script? Can anyone point me to the direction how to solve this issue? Thank you Efelin Kamailio version: kamailio 4.4.0-dev6 (i386/linux) a66e22-dirty

2 2

Kamailio as UAC
by Alexandru Covalschi 27 Nov '15

27 Nov '15

Hello everyone! Subj is an ethernal theme, I know, but odds are so that I just need to do that. I've configured UAC auth to successfully register and my route[PSTN] looks like that # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } if !ds_is_from_list() return; # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } remove_hf("To"); insert_hf("To: <sip:$rU@sipprovider.com>\r\n", "Call-ID"); uac_replace_from("","sip:user@sipprovider.com"); route(RELAY); exit; On INVITE's I get 407 PROXY-AUTH, which are transfered to backend FS. If I try to put if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "Remote asked for authentication"); uac_auth(); } to MANAGE_FAILURE or MANAGE_REPLY route Kamailio can't start. Is that even possible? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 4

how to log sql query
by John Smolka 27 Nov '15

27 Nov '15

Hi guys, I use kamailio with mysql. Some queries are huge and when I need to debug something it takes a time, so the debbuging would be easier. Is there any way how could I log all sql queries into a log file ? Thanks, John

2 1

Media address family
by Sergey Okhapkin 27 Nov '15

27 Nov '15

What is the best way to find out address family (IPV4 or IPV6) of media in SDP? My goal is to reject INVITEs which have different address families in headers and SDP, i.e. IPv4 INVITE with IPv6 media address in SDP and vice versa.

3 2

Correct start up of RTPProxy and Kamailio
by Michael Nielsen 27 Nov '15

27 Nov '15

I've installed RTPProxy and Kamailio and would like to have them start up correct on boot. First RTPProxy and then Kamailio. How to accomplish this?

2 1

2 Kamailio servers on same domain
by Michael Nielsen 26 Nov '15

26 Nov '15

I've installed Kamailio on 2 different servers. They share the same PostgreSQL database for users etc. I've setup a single domain, sip.my-domain.com, with latency responds to the 2 servers. So a user will always land on the server with the fastest responds in a respective area of the world. However, it seems as if 2 users are registered on different Kamailio servers - even though they share the same database - they cannot call each other? Have I misunderstood something? I've enabled WITH_NAT.

2 5

kamailio very slow or stuck during period from time to time
by Thibault Gueslin 26 Nov '15

26 Nov '15

I am facing an issue with kamailio 4.3.3 that is very similar to http://lists.sip-router.org/pipermail/sr-users/2015-May/088195.html Had someone found the solution or faced similar issue ? Sometimes kamailio is receiving the message (check with wireshark, logs in tcp) but the message is not handled nor relayed. After a certain amount of time everything works fine. I will do what as has been advised : enabling pike & check with gdb when it appears. thibault

3 13

issue using sipcapture to pg database
by david escartin 26 Nov '15

26 Nov '15

hello how are you? i have a problem using sipcapture module i'm writing in postgres database some sip messages but the msg field is stored ina strange format, \x5349502f322e30203438372052657175657374205465726d696e61746 5640d0a5669613a205349502f322e302f5544502037392e3137302e36342e3136383a353038303b6272616e63683d7a39684734624b30583665375a70394e446533513b72656365697665643d37392e3137302e36342e3136383b72706f72743d353038300d0a5265636f72642d526f7574653a 203c7369703a3139302e3136362e3233322e3139333a393236323b7472616e73706f72743d7564703b6c723e2c3c7369703a37392e3137302e36382e3138363b6c723b667461673d394448397648374435444653653b6469643d3639612e616437393e0d0a43616c6c2d49443a2033323666333 037632d373138642d346164392d393234362d3333336432323466613439380d0a46726f6d3a2022313931373838393437363922203c7369703a31393137383839343736394037392e3137302e36342e3136383e3b7461673d394448397648374435444653650d0a546f3a203c7369703a313830 39323935373838374037392e3137302e36382e3138363a353036303b757365723d70686f6e653e3b7461673d736263303830367974646c746f6c302d43432d33300d0a435365713a20383338353333383820494e564954450d0a5761726e696e673a203339392031302e31382e352e363420225 353313230303031463135364c3934345b30303030305d2043616e63656c207265636569766564206f6e20696e697469616c20696e76697465220d0a436f6e74656e742d4c656e6774683a20300d0a0d0a do you know how to configure it to see it in ascii? thanks a lot and regards david escartin

2 1

[CFP] reminder! FOSDEM RTC dev-room talks: deadline Friday
by Daniel Pocock 25 Nov '15

25 Nov '15

Reminder: speaker's deadline this Friday, 27 November at 23:59 UTC We have already received several really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the call for papers: http://danielpocock.com/fosdem-2016-free-rtc-dev-room-and-lounge We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

1 0

Kamailio v4.3.4 Released
by Daniel-Constantin Mierla 25 Nov '15

25 Nov '15

Hello, Kamailio SIP Server v4.3.4 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.4. For more details about version 4.3.4 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/11/kamailio-v4-3-4-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

TLS not enough memory issue with git master
by Anthony Messina 25 Nov '15

25 Nov '15

I have noticed the following issue which began with builds somewhere between git master commits bff0a08 and 6173ef7. I did not see this issue with my previous builds and haven't been able to pin down the problem, which is why I haven't formally filed a bug. Any help or guidance is appreciated, because this has crippled my use of Kamailio. Only a restart enables it to work again until the issue recurs. ERROR: tls [tls_server.c:189]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: 8870536 ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading I currently build against and run openssl-1.0.1k-12.fc22.x86_64. I have a very small operation and the only change on the operational side is that all 5 of my mobile UACs (yes, that's all) have switched from CSipSimple/Android to Zoiper/Android, which doesn't yet have support for client-side certificates so verify_certificate and require_certificate are off for both the server and client config. The server is started with: /usr/sbin/kamailio -P /run/kamailio/kamailio.pid -m 64 -M 8 I have tried modifying the shared mem to 128 but the issue still occurs. Even right now, I am seeing the error when only one UAC has established a TLS connection: # kamcmd tls.list { id: 572 timeout: 3475 src_ip: 10.77.79.156 src_port: 58688 dst_ip: 10.77.79.3 dst_port: 5061 cipher: ECDHE-RSA-RC4-SHA SSLv3 Kx=ECDH Au=RSA Enc=RC4(128) Mac=SHA1 ct_wq_size: 0 enc_rd_buf: 0 flags: 2 state: established } # kamailio.cfg enable_tls=yes loadmodule "tls.so" modparam("tls", "connection_timeout", 60) #modparam("tls", "tls_log", 1) #modparam("tls", "tls_debug", 1) #modparam("tls", "low_mem_threshold1", -1) #modparam("tls", "low_mem_threshold2", 0) modparam("tls", "session_cache", 1) # tls.cfg [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/example.org.key.pem certificate = /etc/kamailio/example.org.crt.pem server_name = example.org cipher_list = ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-SHA384:ECDHE-RSA- AES128-GCM-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-RSA-RC4-SHA:ECDHE-RSA-AES256- SHA:DHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES256-SHA256:DHE-RSA-AES128-GCM- SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA:RC4- SHA:AES256-GCM-SHA384:AES256-SHA256:CAMELLIA256-SHA:ECDHE-RSA-AES128- SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:CAMELLIA128- SHA:!aNULL:!eNULL:!EXPORT:!DES:!3DES:!MD5:!PSK [client:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/example.org.key.pem certificate = /etc/kamailio/example.org.crt.pem cipher_list = ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-SHA384:ECDHE-RSA- AES128-GCM-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-RSA-RC4-SHA:ECDHE-RSA-AES256- SHA:DHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES256-SHA256:DHE-RSA-AES128-GCM- SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA:RC4- SHA:AES256-GCM-SHA384:AES256-SHA256:CAMELLIA256-SHA:ECDHE-RSA-AES128- SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:CAMELLIA128- SHA:!aNULL:!eNULL:!EXPORT:!DES:!3DES:!MD5:!PSK Thanks. -Anthony -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

3 16

Kamailio/RTPProxy incorrect port 0
by Igor Potjevlesch 25 Nov '15

25 Nov '15

Hello, I can't figure out why Kamailio reports hundreds of calls with the following error: "/usr/local/sbin/kamailio[2997]: ERROR: rtpproxy [rtpproxy.c:2727]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" It seems that each time this error occurs, we got a one-way audio call. I already check and increase the limit for opening files that the process can open with ulimit -n 500000. But, anyway, if I look to /proc/$PID/fd, I never see more than 1000 files open. I also check the socket with netstat: same; I can see the error while there's only less than 800 sockets open. Is someone know what can be the trigger of this? Regards, Igor.

5 10

Kamailio:Contact validation during INVITE
by Prashant Desai 25 Nov '15

25 Nov '15

Hi, We observe that the following kamailio API is trying to find out user registration by comparing source IP and port with stored contact, if it fails, then it compares based on incoming SIP msg contact uri. is_registered(struct sip_msg* _m, udomain_t* _d){ getContactP(_m,_d); } pcontact_t * getContactP(struct sip_msg* _m, udomain_t* _d) {); //Getting pcontact from source IP and port if (ul.get_pcontact_by_src(_d, &received_host, _m->rcv.src_port, _m->rcv.proto, &c) == 1) { LM_DBG("No entry in usrloc for %.*s:%i (Proto %i) found!\n", received_host.len, received_host.s, _m->rcv.src_port, _m->rcv.proto); } if (c == NULL) { //Getting pcontact from contact uri. if (ul.get_pcontact(_d, &ct->uri, &c) == 0) { } } We have use case where INVITE message is from same source IP and port but having unregistered contact.With this scenario,is_registered() will return success. We would like to know reason behind comparing the source IP and port to fetch the pcontact? Regards, Prashant

3 5

Releasing Kamailio v4.3.4
by Daniel-Constantin Mierla 25 Nov '15

25 Nov '15

Hello, I am considering to release Kamailio v4.3.4 next week on Wed, Nov 25 or the following day. As usual, announce any issues that you are aware and not reported to sr-dev or bug tracker. Also, if some patch was not backported and you think it has to be, 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 Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 1

Build error on Solaris 11 with Solaris Studio
by Spencer Thomason 25 Nov '15

25 Nov '15

Hello, I realize its probably uncharted waters but I was hoping someone could point me in the right direction… I attempting to build Kamailio on a UltraSPARC T2 box running Solaris 11.3 with Sun cc and I’m not getting very far: $ CC=cc gmake all CC (cc) [kamailio] timer_proc.o cc: Warning: -xarch=v9 is deprecated, use -m64 to create 64-bit programs "timer_funcs.h", line 111: warning: argument mismatch "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "cfg/../atomic/atomic_unknown.h", line 59: #warning: no native memory barrier implementations, falling back to slow lock based workarround "cfg/../lock_alloc.h", line 72: warning: argument mismatch "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "ut.h", line 236: warning: argument mismatch "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 240: warning: argument mismatch Any help would be greatly appreciated! The contents of config.mak are below: # this file is autogenerated by make cfg MAIN_NAME= kamailio CFG_NAME= kamailio SCR_NAME= sip-router FLAVOUR= kamailio INSTALL_FLAVOUR= kamailio SRC_NAME= sip-router RELEASE= 4.3.3 OS= solaris ARCH= sparc64 C_DEFS= -DNAME='"kamailio"' -DVERSION='"4.3.3"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"cc 5.1"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=4003003 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DDBG_QM_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DUSE_PTHREAD_MUTEX -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H DEFS_RM= PROFILE= CC= cc LD= cc MKDEP= cc -xM1 MKTAGS= ctags LDFLAGS= -xO5 C_INCLUDES= CC_MKDEP_OPTS= MOD_LDFLAGS= -G -xO5 LIB_LDFLAGS= -G -xO5 UTILS_LDFLAGS= -xO5 LIB_SONAME= "-h " LD_RPATH= -"-R " LIB_SUFFIX= .so LIB_PREFIX= lib LIBS= -lfast -ldl -lresolv -L/usr/local/lib -lxnet -lnsl -lrt LEX= flex YACC= yacc YACC_FLAGS= -d -b cfg PREFIX= /usr/local LOCALBASE= /usr/local TAR= gtar INSTALL= ginstall INSTALL_CFG= ginstall -m 644 INSTALL_BIN= ginstall -m 755 INSTALL_MODULES= ginstall -m 755 INSTALL_DOC= ginstall -m 644 INSTALL_MAN= ginstall -m 644 INSTALL_LIB= ginstall -m 755 INSTALL_TOUCH= touch INSTALL_SHARE= ginstall -m 644 Q= 1 CC_EXTRA_OPTS= CPU= CFLAGS_RM= CFLAGS= -m64 -g -xO5 -fast -native -xarch=v9 -xCC -xc99 $(CC_EXTRA_OPTS) MOD_CFLAGS= -xcode=pic32 $(CFLAGS) LIB_CFLAGS= -xcode=pic32 $(CFLAGS) UTILS_CFLAGS= $(CFLAGS) BASEDIR= $(DESTDIR) basedir= $(BASEDIR) DESTDIR= LIBDIR= lib64 RUNBASEDIR= $(DESTDIR) runbasedir= $(RUNBASEDIR) PREFIX= $(LOCALBASE) prefix= $(PREFIX) cfg_prefix= $(basedir)$(prefix) cfg_dir= etc/$(MAIN_NAME)/ bin_prefix= $(basedir)$(prefix) bin_dir= sbin/ modules_prefix= $(basedir)$(prefix) modules_dir= $(LIBDIR)/$(MAIN_NAME)/ doc_prefix= $(basedir)$(prefix) doc_dir= doc/$(MAIN_NAME)/ man_prefix= $(basedir)$(prefix) man_dir= man/ ut_prefix= $(basedir)$(prefix) ut_dir= share_prefix= $(basedir)$(prefix) share_dir= share/$(MAIN_NAME)/ lib_prefix= $(basedir)$(prefix) lib_dir= $(LIBDIR)/$(MAIN_NAME)/ data_prefix= $(basedir)$(prefix) data_dir= $(MAIN_NAME)/ cfg_target= $(prefix)/$(cfg_dir) lib_target= $(prefix)/$(lib_dir) data_target= $(prefix)/$(data_dir) run_prefix= $(runbasedir) run_dir= var/run/$(MAIN_NAME)/ run_target= $(run_prefix)/$(run_dir) override makefile_defs:=1 C_DEFS:=$(filter-out $(DEFS_RM) $(extra_defs),$(C_DEFS)) $(extra_defs) CFLAGS:=$(filter-out $(CFLAGS_RM) $(CC_EXTRA_OPTS), $(CFLAGS)) $(CC_EXTRA_OPTS) Thanks! Spencer

1 0

Grab users password from WWW-Auth header
by Alexandru Covalschi 24 Nov '15

24 Nov '15

Hello! My problem is I need to do users authentication through API. So I need to replace if (!www_authenticate("$td", "subscriber")) { www_challenge("$td", "1"); } With if (!my_auth_script()) { www_challenge("$td", "1"); } The main problem is - how can I grab or compare users password? I know nonce, which I understand is MD5 salt. Can I, for example, grab users password from API, then grab the MD5 string and the nonce user sent me, calculate MD5 on base of API password and nonce - and then compare MD5 strings sent by user and calculated? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

3 11

Issue with SNMPstats
by Igor Potjevlesch 24 Nov '15

24 Nov '15

Hello, I have configured the SNMPstats module fine in my machine because I get the SNMP information with the new KAMAILIO MIBS but I don't understand why the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 However I can see the transactions: KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32: 54797 Should I configure an specific parametr of the SNMPstats module? Regards,

3 6

MESSAGE is not successfull in Kamailio IMS
by Thierry Luo 24 Nov '15

24 Nov '15

Hi All, I installed kamailio on Ubuntu as IMS pCSCF/iCSCF/sCSCF. REGISTER is fine, but when I tried to send a MESSAGE (SMS) from one user to another, it was always not successful. I captured IP packets on the server, and it seemed that the error was caused on pCSCF. User1 and User2 are both behind their NAT network. Suppose User1's private IP is Private_IP1, and User1's public IP (on router) is Public_IP1, and User2's private and public IPs are Private_IP2 and Public_IP2: User1 pCSCF iCSCF sCSCF user2 MESSAGE ----------------> MESSAGE ---------------------> MESSAGE -------------------> MESSAGE <----------------------------------------------------- MESSAGE ---------------------------------------------------------------------------- ----> Unfortunately, from the captured data on the server, in the last step, the MESSAGE was sent to the Private_IP2 of User2. It should be sent to the Public_IP2. Of course, the datagram with the destination of Private_IP2 cannot reach User2 successfully. Please help analyze why the pCSCF sent the MESSAGE packet to User2's private IP instead of public IP. The configuration file was attached for reference. BR. Thierry Luo

2 6

Simple User Agent change
by Steven Maddox 24 Nov '15

24 Nov '15

Hi, Sorry to ask something probably very simple but I'm difficulty finding the right kinds of resources to learn from which are succinct enough to not overload my head. We have a PBX which we don't control (and hate) which expects only certain makes/models phones to register (it checks for user agents). I have a phone which unfortunately doesn't support tweaking its user agent, so I'd like to pass it through an extremely SIP proxy to have it's user agent tweaked and all other data to/from the PBX transparently let through. I've managed to knock up something (that isn't a proxy) using Asterisk where I made a trunk and a peer... but obviously since this isn't a proxy... things like BLF and transferring calls will never work. If anyone can point me in the direction of a really simple set up to achieve this, please let me know :) -- Steven Maddox

2 2

kazoo module documentation
by Vik Killa 24 Nov '15

24 Nov '15

Hello, I'm testing kazoo module for use with RabbitMQ I noticed there is a required parameter that is not listed in the documentation: http://kamailio.org/docs/modules/devel/modules/kazoo modparam("kazoo", "pua_mode", 0) The above parameter is required but not documented. Can someone please update the documentation? Thank you. /V

2 1

Problem with resuming suspended transactions.
by Dmytro Bogovych 23 Nov '15

23 Nov '15

Greetings. We build specialized softphone for iOS. Push notifications are used to notify user and wake up softphone app on missed calls and messages. We used Daniel's example config script as base for own configuration file. It has the same parts and does not differ too much. I talk about this example: http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… Everything was ok for few monthes - with kamailio 4.1.2. Few weeks ago we upgraded to 4.3.3. Now paused calls cannot be resumed properly. Softphone receives 500 Server error (No error 0/SL) in response This is problem part of script: route[RELAYRESUMED] { # 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 ($var(rr_target) != "") { xlog("L_INFO","Relay to $var(rr_ip):$var(rr_port)\n"); $ru = "sip:" + $rU + "@" + $var(rr_ip) + ":" + $var(rr_port) + ";transport=" + $var(rr_proto); xlog("L_INFO","Current request URI is $ru\n"); if (!t_relay()) xlog("L_INFO","Cannot relay PUSHED invite."); sl_reply_error(); } else { xlog("L_WARN","Target proto is not set\n"); sl_reply_error(); } exit; } It produces log ----- Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: resuming trasaction [23927:345343979] 1985987629 (23927:345343979) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Relay to 95.133.110.17:11791 Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Current request URI is sip:1985987629@95.133.110.17:11791;transport=tls Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (0/SL) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Checking FAILED messages for 1985987629: 0 ----- sl_reply_error() sends 500 with No error (0/SL) description. Later 180 & 200 are sent. But client softphone terminates transaction already (as it handles 500 response code) . There is no "Cannot relay PUSHED invite." message in the log however. I will be happy to heary any idea what can be cause... Thank you!

2 3

Kamailio-SEMS Integration
by Prashant Desai 23 Nov '15

23 Nov '15

Hi, We are working on kamailio-4.2.3 and we require B2BUA functionality to be achieved.From kamailio official website we got to know,we can integrate kamailio with SEMS to achieve B2BUA functionality. We have few queries regarding this integration. 1.SEMS-1.6.0(latest version available) is been used,is this version suits kamailio-4.2.3? 2.SEMS-1.6.0 does not support unix based communication with kamailio-4.2.3,as SEMS-1.6.0 doesn't have unix socket interface plugin. Since we need unix socket based communication with SEMS.can we port the unix socket interface plugin to SEMS-1.6.0?If no,please suggest us better approaches. Regards, Prashant.

2 2

Kamailio, rtpproxy, websocket and 488 error
by Alfredo Pastor 23 Nov '15

23 Nov '15

Hello, I was doing some tests with Elastix MT that use version 4.1.6 of Kamailio and by default uses rtpproxy, I had the problem of error 488 and unable to make calls from websockets (Incompatible SDP). I could see that Peter Dunkley in a video explains that the problem is solved by rewriting the SDP, but he uses the mediaproxy not rtpproxy. What are your recommendations? I hope you can help me. Chreers, Alfredo.

1 0

rtpengine via-branch=extra question
by Juha Heinanen 22 Nov '15

22 Nov '15

rtpengine readme tells: via-branch=... - Include the “branch” value of one of the “Via” headers in the request to the RTP proxy. Possible values are: “1” - use the first “Via” header; “2” - use the second “Via” header; “auto” - use the first “Via” header if this is a request, or the second one if this is a reply; “extra” - don't take the value from a header, but instead use the value of the “extra_id_pv” variable. This can be used to create one media session per branch on the RTP proxy. When sending a subsequent “delete” command to the RTP proxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “rtpengine_delete”, or stop all sessions for a call when not passing one of those two flags there. if via-branch=extra is used in offers of branched, should it also be used in answers and deletes? -- juha

2 6

Functionality of rtpproxy
by kai.ohnacker＠cbc.de 20 Nov '15

20 Nov '15

Hello, I have installed a kamailio server with rtpproxy. The server is located between 2 networks and bridge these networks. My question: Does handle rtpproxy only 1 call at the same time? Because my configuration works, but only for one call from the first to the second network. The second call is not working. But if I make internal or external call, I can do more than one call. Do I need the multiple rtpproxy usage configuration? I thought not, because I don't use different IPs for the second call... Cheers, Kai

2 1

403 Forbidden for INVITE in kamailio for ims call
by Priyaranjan Nayak 20 Nov '15

20 Nov '15

Hi All, I have installed kamailio and using as a i-cscf,s-cscf and p-cscf server. I have taken FHoSS(OpenImscore) for hss server Now I am able to do register and unregister. When I am sending INVITE for call, the calling-party and called-party were registerd but I am getting "403 Forbidden - Dialog not found on S-CSCF or Terminating user not suitable for unregistered services". Could you please tell me what is the root cause of above error message ? Best regards, Priyaranjan

1 0

RFC: rtpproxy and a=nortpproxy:yes
by Daniel-Constantin Mierla 19 Nov '15

19 Nov '15

Hello, I am looking to get some feedback on some issues I noticed more and more lately. Apparently some SIP devices (media servers, phones, ...) are keeping the "a=nortpproxy:yes" line in SDP when replying to an INVITE that contains such line. [Alice] ------> [Kamailio+RTPProxy] ------> [Bob] The 200ok response from Bob has "a=nortpproxy:yes" in SDP. By default, that line in SDP makes the rtpproxy not to engage itself anymore in rtp relaying, and as a result things like no audio or one way audio happens. Anyone else encountering such situations? If yes, what are the devices with such behaviour? So far I noticed with some FreeSwitch and Snom -- none of them I can control, so there might be a specific configuration of those devices, not something by default there. The solution is to set: modparam("rtpproxy", "nortpproxy_str", "") and use flag 'r' for rtpproxy_manage() if the IP in SDP is not a private address. I already updated the default config for master to use flag 'r' if the SDP media IP is not private, wondering if nortpproxy_str should be set to empty in kamailio.cfg (or made empty as default in config). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

rtpproxy database structure for kamailio 4.2.3
by Jan Hazenberg 19 Nov '15

19 Nov '15

Hi all, I'm trying to load the rtpproxy url's from database but get a error when starting kamailio (4.2.3): ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'set_name' in 'field list' Looks like the field set_name is missing in the database structure, but i can't find it in the docs. If i add the field kamailio starts, but what is the field set_name used for. Seems it needs to be a int value: ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name 'test' - must be integer Thanks, Jan

2 2

Fw: Kamailio with Siremis
by Ryan Holbein 19 Nov '15

19 Nov '15

________________________________ From: Ryan Holbein <rtholbein(a)hotmail.com> Sent: Wednesday, November 18, 2015 8:07 AM To: miconda(a)gmail.com Subject: Re: [SR-Users] Kamailio with Siremis Hello, This is what I have setup so far I know it is not much but looking for help this is my very first Proxy. If you look at my Attachments I am trying to do IP auth. I have 2 Asterisk Boxs with 192.168.89.2 and .3.. The Proxy has 2 nics the outside and the 192.168.89.0/24 ... How do I setup the LCR rule if you look at the LCR Attachment? Also is my trusted permissions setup correct(Those IPs you see are correct)? If you are willing to help Thank you in Adv. ________________________________ From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Thursday, November 12, 2015 12:05 PM To: Ryan Holbein; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis If you have a single Asterisk box, then you can simply forward using kamailio.cfg only. Look at route[PSTN] in default configuration file for an example. If you have more Asterisk boxes, then depends on what you need/want to do: - for load balancing look at dispatcher module - it has a complete sample config in the docs - for least cost routing, then lcr module is indeed an option Cheers, Daniel On 12/11/15 17:56, Ryan Holbein wrote: Now I assume in LCR I will add the Asterisk boxs so it can route? Does that sound correct? Anyone able to provide an example? Thank you ________________________________ From: sr-users <sr-users-bounces(a)lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org> on behalf of Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com> Sent: Thursday, November 12, 2015 3:01 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis Hello, look at permissions module. Default kamailio can do IP auth by enabling WITH_AUTH and WITH_IPAUTH and adding the trusted IP in database table 'address' with group id 1. Siremis has a panel for managing the records in address table. Cheers, Daniel On 11/11/15 14:10, Ryan Holbein wrote: Hello, I have setup with my phone provider an IP endpoint to do IP Auth. How would I go about setting that up in Kamailio or even better can some tell me how to do that in Siremis? _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda<http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

2 2

ndb_mongodb, db_mongodb and replicasets
by Kelvin Chua 19 Nov '15

19 Nov '15

anybody ever tried using a mongodb url with replicasets and parameters to work with kamailio? Kelvin Chua

2 11

Content-Type
by Safdar Khan 19 Nov '15

19 Nov '15

How to check content-type of ongoing various type of media streams in kamailio?

2 1

siptrace segfault - attack?
by Marian Piater 18 Nov '15

18 Nov '15

Hello, we had segfault today. Kamailio has been running for few months without problems. Nov 18 10:51:20 sbc kernel: [11326028.926502] kamailio[14452]: segfault at 30 ip 00007f20f7f3838a sp 00007ffef3ab7b10 error 4 in siptrace.so[7f20f7f23000+27000] GDB: Reading symbols from kamailio...done. [New LWP 14452] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kam'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 1041 sto.totag = get_to(msg)->tag_value; (gdb) bt #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 #1 0x0000000000457ca9 in do_action (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1053 #2 0x0000000000463cb9 in run_actions (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1548 #3 0x00000000004643bf in run_top_route (a=0x7f2100b8a928, msg=0x7f2100f96750, c=0x0) at action.c:1634 #4 0x0000000000573ea7 in receive_msg ( buf=0x9c9400 <buf> "OPTIONS sip:100@XXX.XXX.XX.XX SIP/2.0\r\nv: SIP/2.0/UDP 69.64.39.119:5060;branch=z9hG4bK-82135822;rport\r\nContent-Length: 0\r\nf: \"MisterX\"<sip:100@1.1.1.1>;tag=61326665333131663133633401333733313630343335"..., len=357, rcv_info=0x7ffef3ab85b0) at receive.c:196 #5 0x0000000000493ff5 in udp_rcv_loop () at udp_server.c:495 #6 0x000000000051fdd7 in main_loop () at main.c:1573 #7 0x0000000000525b6b in main (argc=13, argv=0x7ffef3ab8998) at main.c:2533 (gdb) quit Our IP is hidden, but there is 69.64.39.119, which is foreign address and I think, it was attack. Unfortunatelly I don't have SIP packet details, but you can see 1.1.1.1 or strange tag or content-length=0 in received message. I looked into siptrace.c and there is function sip_trace_prepare where is get_from(msg) checked, but no get_to(msg). This function is run from main sip_trace function. So I think, we need also check get_to(msg) function. I just disabled siptrace module, but we need it. Thank you. Marian

1 1

Stop msrp relay from kamailio to freeswitch
by Safdar Khan 18 Nov '15

18 Nov '15

Hi folks I have integrated kamailio and freeswitch. The basic register and authentication is happening through kamailio and other media services through freeswitch like file sharing (using msrp). My question is how can I stop file sharing through msrp via freeswitch since everything going from kamailio to freeswitch to client. I just wanted to share file only through kamailio. I am attaching my cfg file below... #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_PRESENCE #!define WITH_TLS #!define WITH_MSRP #!define WITH_XHTTP #!define WITH_MSILO #!define WITH_NAT #!substdef "!MY_IP_ADDR!192.168.5.99!g" #!substdef "!MY_DOMAIN!<SERVER_IP>!g" #!substdef "!MY_WS_PORT!9080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!define WITH_WEBSOCKETS # Kamailio (OpenSER) SIP Server v4.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:127.0.0.1:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_FREESWITCH freeswitch.bindip = "192.168.5.99" desc "FreeSWITCH IP Address" freeswitch.bindport = "5090" desc "FreeSWITCH Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/kamailio-4.3/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_MSRP loadmodule "msrp.so" #!endif loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "cfgutils.so" #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" ##loadmodule "nathelper.so" #!endif #!ifdef WITH_MSILO loadmodule "msilo.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- #modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.5.99:5060") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "NULL") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-4.3/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/usr/local/kamailio-4.3/etc/kamailio/privkey.pem") modparam("tls", "certificate", "/usr/local/kamailio-4.3/etc/kamailio/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/kamailio-4.3/etc/kamailio/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) modparam("tls", "crl", "/usr/local/kamailio-4.3/etc/kamailio/my_crl.pem") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif #!ifdef WITH_MSRP # ----- msrp params ----- modparam("msrp", "sipmsg", 1) modparam("msrp", "cmap_size", 8) modparam("msrp", "use_path_addr", "msrp://192.168.5.99:5060") #!endif modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=8") modparam("sqlops","sqlcon","ca=>mysql://kamailio:kamailiorw@localhost/kamailio") #!ifdef WITH_AUTH # ----- auth params ----- #modparam("auth", "nonce_count", 1) #modparam("auth", "qop", "") #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif #!ifdef WITH_MSILO # ------ msilo params ------ modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@192.68.5.98") modparam("msilo","contact_hdr","Contact: <sip:registrar@192.168.5.99>\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","") modparam("msilo", "check_time", 200000) modparam("msilo", "expire_time", 259200) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE_MESSAGE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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 route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); ## #!ifdef WITH_MSILO ### m_dump(); ### #!endif exit; } } # USER location service route[LOCATION] { xlog("going inside LOCATION\n"); #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_MSILO xlog("going inside ifdefMSILO\n"); #if(is_method("MESSAGE")) # { xlog("going inside MSILO\n"); route(MSILO); # exit; #} #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif #!ifdef WITH_MSILO if(is_method("MESSAGE")){ return; } #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); route(RELAY); exit; } #!endif # manage outgoing branches # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_MSILO #manage failure message failure_route[MANAGE_FAILURE_MESSAGE] { if (!method=="MESSAGE") { exit; } log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { t_reply("202", "Accepted"); }else{ t_reply("503", "Service Unavailable"); } } #!endif # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); route(FSVBOX); } } #!endif route[8] { sql_query("ca", "insert into statistics (time_stamp,random,shm_used_size,shm_real_used_size, shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts, $RANDOM,$stat(used_size),$stat(real_used_size),$stat(max_used_size), $stat(free_size),$stat(location-users),$stat(location-contacts))", "ra"); } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0", "$msrp(method)")) { if(auth_get_www_authenticate("myrealm", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } msrp_cmap_save(); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } if (msrp_cmap_lookup()) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_relay(); } else { msrp_reply("481", "Session-does-not-exist"); } } else { msrp_reply("501", "Request-method-not-understood"); } } #--------------websockets starts here------------- #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; exit; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif #--------------websockets ends here------------- #---------------msilo starts here------------------ route[MSILO]{ #!ifdef WITH_MSILO if(!is_method("MESSAGE")) return; ##..........added...... if(lookup("location")) { if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; }; if(!lookup("location")) { $var(rc)=$rc; if (! t_newtran()) { sl_reply_error(); exit; }; switch ($var(rc)) { case -1: case -3: if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; case -2: t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; } }; t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; #!endif return; } #------------------msilo ends here----------------

1 0

Kamailo crashes with auth_radius module
by Volkan Oransoy 18 Nov '15

18 Nov '15

Hi, I try to enable auth_radius module with my kamailo box. But it crashes on the first REGISTER request. I have enabled it with this two lines. loadmodule "auth_radius.so" modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") This is the authentication part of my script. if (is_method("REGISTER")) { if (!radius_www_authorize("$td")) { switch ($rc) { case -7: send_reply("500", "Server Internal Error"); exit; case -1: send_reply("400", "Bad Request"); exit; default: }; if (defined($avp(digest_challenge)) && ($avp(digest_challenge) != "")) { append_to_reply("$avp(digest_challenge)"); }; send_reply("401", "Unauthorized"); exit; }; } And in the /etc/radiusclient/dictionary file, I included sip and kamailio dictionaries. $INCLUDE dictionary.sip $INCLUDE dictionary.kamailio Service succesfully starts but it crashes on the first REGISTER request. Nov 18 11:30:27 sip kernel: [159645.360105] traps: kamailio[6941] general protection ip:7f89b1660920 sp:7ffc8d3bac78 error:0 in libc-2.19.so [7f89b15cb000+19f000] Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:728]: handle_sigs(): child process 6941 exited by a signal 11 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:731]: handle_sigs(): core was generated Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6959]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6951]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6936]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6948]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6944]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6937]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6953]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6945]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6938]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6955]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6943]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6942]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6957]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6939]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6940]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Thanks,

2 3

Authorization selects
by Alexandru Covalschi 17 Nov '15

17 Nov '15

Hello everyone! I need to extract values from authentication header, but 408. @authorization["string"] 409. @authorization["string"].username 410. @authorization["string"].username.user 411. @authorization["string"].username.domain 412. @authorization["string"].realm 413. @authorization["string"].nonce 414. @authorization["string"].uri 415. @authorization["string"].cnonce 416. @authorization["string"].nc 417. @authorization["string"].response 418. @authorization["string"].opaque 419. @authorization["string"].algorithm 420. @authorization["string"].qop these values doesn't seem to work for me. Used https://github.com/kamailio/kamailio/blob/master/doc/select_list/select_cor… doc. $var(z) = @authorization["string"].realm; xlog("Realm $var(z)"); Using that simple check I get nothing in $var(z) version: kamailio 4.3.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2 Can anyone check that please? Or maybe syntax is wrong and I need newer selects doc? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 3

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

4 8

General question about mohqueue
by David Villasmil Govea 16 Nov '15

16 Nov '15

Hello list, I've been given a task to setup a predictive dialer with for 8000+ agents taking accepted calls in a queue. I've done this for a much smaller setup in freeSWITCH but i recently saw that kamailio has a mohqueue (which i already setup as a PoC) that might work for this. My question is whether you think it's a good idea to go with this on kamailio or is freeSWITCH better suited for this kind of application? The setup would be to generate outgoing calls based on a list (predictive dialer) and send answered calls to a queue to be answered by an agent. Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Kamailio 4.3 and Debian 8.2 systemctl
by Grant Bagdasarian 16 Nov '15

16 Nov '15

Hello, I've just installed Kamailio 4.3 on Debian 8.2, but starting kamailio does not work since Debian requires systemctl to be setup. I've tried the below config but it didn't work (https://github.com/kamailio/kamailio/issues/294#issuecomment-138590939) usr/local/src/kamailio# systemctl status kamailio.service â kamailio.service - Kamailio SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled) Active: failed (Result: resources) systemd[1]: [/etc/systemd/system/kamailio.service:14] Not an absolute path, ignoring: $PIDFILE systemd[1]: Failed to load environment files: No such file or directory systemd[1]: kamailio.service failed to run 'start' task: No such file or directory systemd[1]: Failed to start Kamailio SIP Server. systemd[1]: Unit kamailio.service entered failed state. [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=syslog.target network.target [Service] Type=forking Environment='PIDFILE=/var/run/kamailio/kamailio.pid' Environment='CFGFILE=/etc/kamailio/kamailio.cfg' Environment='SHM_MEMORY=64' Environment='PKG_MEMORY=8' Environment='USER=kamailio' Environment='GROUP=kamailio' EnvironmentFile=/etc/default/kamailio PIDFile=$PIDFILE # ExecStart requires a full absolute path ExecStart=/usr/sbin/kamailio -P $PIDFILE -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP ExecStopPost=/bin/rm -f $PIDFILE Restart=on-abort [Install] WantedBy=multi-user.target Does anyone have a working kamailio.service file? Regards, Grant

4 3

TLS module not initialized in 4.3.3, worked in 4.1.5
by Sebastian Damm 16 Nov '15

16 Nov '15

Hello, we just updated one kamailio server from 4.1.5 to 4.3.3, and although the config file is correct and kamailio starts up, it doesn't initialize TLS and says " tls support enabled, but no tls engine available (forgot to load the tls module?)" In the log I see: Old shutdown (last lines): Nov 13 11:44:38 lasola /usr/sbin/kamailio[15890]: DEBUG: <core> [mem/shm_mem.c:235]: shm_mem_destroy(): destroying the shared memory lock Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:271]: tcp_read_data(): error reading: Connection reset by peer (104) Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:1296]: tcp_read_req(): ERROR: tcp_read_req: error reading New startup (first lines): Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [daemonize.c:583]: set_core_dump(): core dump limits set to 18446744073709551615 Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2475]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2476]: main(): disabling tls... Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [async_task.c:88]: async_task_init(): start initializing asynk task framework Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): xmlrpc Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules//sl.so] Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): sl In our config file we have the following lines for TLS (pretty late, after all other module loading and after most parameters): #!ifdef ENABLETLS loadmodule "tls.so" modparam("tls", "private_key", "/etc/ssl/private/my.kamailio-key.pem") modparam("tls", "certificate", "/etc/ssl/certs/my.kamailio.crt") #!ifdef TLS_CA_CHAIN # Maybe we want to use a chain to the CA modparam("tls", "ca_list", "/etc/ssl/certs/my.ca-bundle.crt") #!endif enable_tls=1 listen=tls:1.2.3.4:5061 #!endif After starting up, kamailio listens on port 5060, but not on port 5061. In version 4.1.1, this config worked without a problem. Has anybody seen this before? the tls module is there and available, it doesn't say anything about "cannot load module", and it is only a warning message. I'm also wondering, why this message is the first after starting the server. From config I would expect that sl, tm and all the other modules should be initialized before tls. Best Regards, Sebastian

2 9

Detecting problems of kamailio under very high load
by aft 16 Nov '15

16 Nov '15

We have multiple kamailio servers with 4 cpu cores and 16G RAM. We use kamailio+rtpproxy as a outbound sip proxy. Usually there are many thousands of concurrent sip sessions of occurring there. Periodically sometimes it just stops serving request and spits out 5** replies. At that point we usually reload the kamailio daemon and things return back to normal. As we have a lot of servers, manually doing this is a pain in the neck. So We installed a homer/sipcapture on a separate server . And from there, we periodically scan for 500 msgs for a given time interval. If there are any, we then proceed to reload kamailio. I know its hell of an inefficient system to monitor. So I'm wondering, if there can be any more rational way to detect problems pre-emptively. any fifo/mi command which we could run to find out if the kamailio instance is hung up? Cheers Arif

3 2

Loading mohqueue stops kamailio from starting up
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello all, I installed kamailio on debian from kamailio's apt repository. Configured it with kamdbctl, started it up properly, no issues. I configured mysql also, worked properly. Next i loaded up "mohqueue" and everything went to hell :) This is my config: loadmodule "mohqueue.so" # ----- moh-queue ----- modparam ("mohqueue", "db_url", "mysql://user:password@localhost/kamailio") modparam ("mohqueue", "db_qtable", "mohqueue") modparam ("mohqueue", "db_ctable", "mohqcalls") modparam ("mohqueue", "mohdir", "/var/kamailio/MOH") and my db table for mohqueue is: +----+------------+-------------------------+--------------------+---------+-------+ | id | name | uri | mohdir | mohfile | debug | +----+------------+-------------------------+--------------------+---------+-------+ | 1 | test_queue | sip:1234@172.16.163.130 | /var/kamailio/MOH/ | test | 9 | +----+------------+-------------------------+--------------------+---------+-------+ when starting, i get: ... 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(5069) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(5069) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(5069) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Anyone can help? Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 2

(no subject)
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello guys, I'm trying to startup 4.3 and i'm getting the following: 0(4420) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 192 bytes for row values at 0x7fe6bf6b5a48 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [1] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [ sip:1234@172.16.163.130] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [/var/kamailio/MOH/] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test.8] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test_queue] 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [9] 0(4420) ERROR: mohqueue [mohq_db.c:504]: update_mohq_lst(): Queue,Field (test_queue,mohdir): Unable to find MOH files (/var/kamailio/MOH//test.8.*)! 0(4420) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules/sl.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_tm> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_newtran> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_tcp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_udp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(4420) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(4420) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(4420) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Any help is appreciated, Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Kamailio queues
by David Villasmil Govea 13 Nov '15

13 Nov '15

Hello Guys, I'm new to this list. I'm trying to setup a queue with 4.3.3 (x86_64/linux) and I see a couple of things i'm wondering about: - kamdbctl created the table "mohqueues" and the associated value in the "version" tables. Whereas kamailio uses "mohqueue" (no extra "s" at the end). - after fixing that, it finds the table, but when searching for the moh file, which i set as "/var/kamailio/MOH/test.8" on the mohqueue table, but kamailio can't find it: 0(61540) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(61540) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules ls -l /var/kamailio/MOH/test.8 -rw-rw-rw- 1 root root 166038 Jun 10 2009 /var/kamailio/MOH/test.8 Can anyone lend a hand? Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Re: [SR-Users] Flaky & unstable RTP connections
by Infinicalls Infinicalls 13 Nov '15

13 Nov '15

Hi, I have started RTP with # ./rtpproxy -F -l localhost -s udp:*:7890 and when I issue #netstat -pln | grep rtpp I get, udp 0 0 0.0.0.0:7890 0.0.0.0:* 8759/rtpproxy I am using another similar instance of RTP also. Still my connection seems to be flaky. This doesn't occur always. I am not using dispatcher here. Simple rtpproxy with udp:rtp1 and udp:rtp2 in kamailio.cfg file. Is there anyway to tell if RTP is healthy other than making calls and finding out? Can somebody test it from your end? Or is this problem client specific? I am using microlite and sipper on windows. phone.infinicalls.com u:test;p:test123 u:trial;p:trial123 5060 regards Ganesh Kumar

1 0

Support-for-AKAv2-MD5-authentication-in-kamailio
by Anil Kumar 13 Nov '15

13 Nov '15

Hi, I want to enable AKAv2-MD5 type authentication using a plain kamailio. Is there any way I can achieve it ? Please do needful. Thanks Anil

2 1

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