- sr-users - kamailio.org

Issue dumping an huge htable through xmlrpc
by Marco Capetta 24 Nov '17

24 Nov '17

Hello everyone, On Kamailio version 4.4.6 I have an issue dumping an huge htable through xmlrpc. We are using htables to store the subscriber registration attempts to prevent bruteforcing password attacks. Usually there aren't any issues to dump information using xmlrpc. On the contrary, if the number of records in the htable become huge (in the order of 10.000), xmlrpc output is truncated, therefore it is not possible to parse it. I have the same results either using a Perl library or the "xmlrpc_test.py" script included in package. I tried to increase the module parameters "binrpc_max_body_size" and "binrpc_struct_max_body_size" without success. Do you have any other suggestion? Thank you Regards Marco -- *Marco Capetta * Operations Engineer Sipwise GmbH <http://www.sipwise.com> , Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Phone: +43(0)1 301 2044 <tel:+4313012044> Email: mcapetta(a)sipwise.com <mailto:mcapetta@sipwise.com> Website: www.sipwise.com <http://www.sipwise.com> Particulars according Austrian Companies Code paragraph 14 "Sipwise GmbH" - Europaring F15 - 2345 Brunn am Gebirge FN:305595f, Commercial Court Vienna, ATU64002206

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Failed to find command http_client_query
by Alexandru Covalschi 24 Nov '17

24 Nov '17

Hello everyone, Updated to 5.0.4 from repo, up-to-date debian 8, http_client.so is certainly loaded. When I try to change http_query to http_client_query it fails to load with error "failed to find command http_client_query". Any ideas? It is same on all my test servers. ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493

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msilo support for picture, audio and video?
by sscc 24 Nov '17

24 Nov '17

Hello Folks does msilo support picture, audio and video? regards sscc? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Loadbalancing authenticated agetways
by SamyGo 23 Nov '17

23 Nov '17

Hi, I've a scenario where multiple mediaservers will be registered to Kamailio and for any incoming call from Upstream provider I've to perform load-balancing to the actively registered media-servers. dispatcher module does load-balancing for IP endpoints, is there any other module where I can perform load-balancing on registered clients(media-servers)? Thanks, Sammy

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why do I need rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk via Kamailio, and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Thank you

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rfc: clang format the code
by Daniel-Constantin Mierla 23 Nov '17

23 Nov '17

Hello, it was discussed during the last irc devel meeting and everyone there agreed to use clang-format to format the source code: - https://clang.llvm.org/docs/ClangFormat.html Is any developer (maybe not present during the irc devel meeting) opposing this? It is not going to be all at once, because some of the structures and initialization of them may be preferred to be kept with a custom format for being easier to read -- to have a section in the file not formatted, it has to be enclosed in between comments: /* clang-format off */ ... /* clang-format o. */ But for the rest of the code (i.e., functions) I think it will be better to agree on a format and try to use it. The clang-formatting should be done when one needs to update code in some files. Over the years, developers come and go, having each one with a custom formatting style is hard to maintain properly in long term. There are modules that do not indent coherently code inside blocks, using a mixture, more or less no style. This makes it hard for other developers to review in case of reported issues, but also for new comers to get familiar with the code. In the source tree there is already a .clang-format file which defines the options for formatting. It was used by me and other developers already -- I did it mainly in my modules or those components that do not have an active maintainer. But I plan to do it also on the components that have currently a maintainer at the moment I need to work on that code (e.g., fix some issue or export functions to kemi framework). My process of clang-formatting a module is like: - edit the files to add clang-format off/on comments around the parts I want to preserve as they are (typically mod exports -- params, functions, kemi -- and intialization of structures that are aligned to a specific indentation that makes it easier to read). - run: clang-format -i src/modules/modname/*.{c,h} - commit with these changes, no new feature or bug fix must be done in the same commit (can be combined with removing the history comments) If one wants different options inside .clang-format, please propose and let's discuss which one is preferred by the majority of the current developers. If no one shows anything against in a matter of few days, it will be considered accepted (after all, one can re-indent on own style if the rule is not accepted or overturned). Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Error when attempting to use rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

Hello, I entered this question before but did not include the tcpdump file. I am re-posting just in case this message was over looked for lack of information. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. My server name is vadev2 (Kamailio run on this server), and it's IP Address and port are 172.21.1.108:5060; Asterisk servers are (172.21.1.97:5060, and 172.21.1.124:5060). The rtpproxy is 172.21.1.108:17722 Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 vadev2/usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.21.1.108:59794<http://172.1.1.13:59794> failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work, other than I do not believe anything is going through rtpproxy; I can make calls and use sip tools which pass requests through Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what and why. Should I be using rtpengine instead of rtpproxy? I did attempt rtpengine but it would not compile due to missing json.h files. Thank you for any help -Steve

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MSILO with encryption/decryption
by SamyGo 23 Nov '17

23 Nov '17

Hi, Is there any internal mechanism for msilo to store messages encrypted and retrieve them decrypted before sending? Thanks, Sammy

1 0

RV: CANCEL not being matched by tm module when via address does not have port included
by CARLOS GARCIA MORCHON 23 Nov '17

23 Nov '17

Hi, I have an IMS core deployed using kamailio 5.0.2. Calls from clients are reaching the core pcscf from a freeswitch (with address A.A.A.A) + a kamailio acting as proxy (with address B.B.B.B), so the INVITE request looks like this: INVITE sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Record-Route: <sip:A.A.A.A;lr> Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Via: SIP/2.0/UDP B.B.B.B:5080;received=B.B.B.B;rport=5080;branch=z9hG4bKHpB704ZaXSKtm Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 INVITE Contact: <sip:mod_sofia@B.B.B.B:5080> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 224 P-Charge-Info: <sip:222222222@mobile.tuenti.int>;npi=ISDN;noa=4 X-FS-Support: update_display,send_info Remote-Party-ID: "222222222" <sip:222222222@B.B.B.B>;party=calling;screen=yes;privacy=off If the client cancels the request, the CANCEL request that gets to the core pcscf looks like this: CANCEL sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 CANCEL Content-Length: 0 Reason: Q.850;cause=16;text="Normal Call Clearing" The processing of the CANCEL at the pcscf is eventually reaching the t_check_trans() function, but the logs return “no CANCEL matching found!” and thus the processing ends. I have followed the code up to the via_matching function in the t_lookup.c file of the tm module source and added some logs. The via matching is failing at the port comparison (the port stored for the INVITE is “5060”, the port stored for the CANCEL is “0”. The flow works ok with a different client (connected directly to the core) which includes the port in the Via headers. ¿Can this be a bug in kamailio or am I doing something wrong?. AFAIK The RFC 3261 does not mandate to include the ports in the sent-by element of the Via header Thanks! Carlos ________________________________ Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it. Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

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Kamailio new install, unable to create users
by Atux Atux 23 Nov '17

23 Nov '17

Hi. New to the area and trying to find my way around with kamailio. In one of my debian servers i need to have kamailio with a gui to start playing around with it. i am following the http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour to install kamailio. When running kamdbctl create it asks what format coding that i must have to configure the DB. i simply add latin1 and it goes on. then when i issue kamctl add daniel 1234qwet it comes with the following error: root@debian:~# kamctl add daniel 1234qwet MySQL password for user 'kamailio@localhost': ERROR 1045 (28000): Access denied for user 'kamailio'@'localhost' (using password: YES) ERROR: introducing the new user 'daniel' to the database failed root@debian:~# Any ideas on how to proceed with the following error please?

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kemi with lua and dispatcher - attempt to index field 'dispatcher' (a nil value)
by Karsten Horsmann 23 Nov '17

23 Nov '17

Hello Daniel, hello List, after reading the fresh released kemi documentation (Thanks for that!) on github i start to configure an kemi lua configuration. It seems to be that there is not an dispatcher export to kemi in my installation. Or i missconfigured something? How to use dispatcher with kemi? 1.) First try to solve this - use param "register" "modulename" to app_lua - doesnt work 2.) grep around in the auto kemi in/export messages in debug startup - nothing to see. 3.) grep around in the kamcmd api.list command for lua - no dispatcher here the intressting parts of the configs and lua-script: kamailio.cfg ... loadmodule "dispatcher.so" loadmodule "permissions.so" loadmodule "avpops.so" #!ifdef WITH_CFGLUA loadmodule "app_lua.so" modparam("app_lua", "reload", 1) modparam("app_lua", "register", "dispatcher") #!endif #!ifdef WITH_CFGLUA modparam("app_lua", "load", "/etc/kamailio/lua/kamailio-kemi-lua.lua") cfgengine "lua" #!else import_file "request-route.cfg" #!endif ... kamailio-kemi-lua.lua function ksr_request_route() ... -- dispatch destinations if KSR.permissions.allow_source_address(FLT_EDGE) then KSR.info("new $fU -> $rU in permissions"); direction = "inbound"; routing_target = FLT_FSNODE; ksr_route_dispatch(direction, routing_target); KSR.x.exit(); end KSR.info("NOT-OKAY - $rU dropped"); -- End of Route() KSR.sl.sl_send_reply(407, "Proxy Authentication Required. See IP GUI"); return 1; ... end --- dispatcher route function ksr_route_dispatch(direction, routing_target) KSR.xlog.xinfo("new $fU -> $rU in routing_target ksr_route_dispatch"); KSR.info(routing_target); if (routing_target == nil) then KSR.xlog.xinfo("routing_target nil"); KSR.sl.sl_send_reply(604, "Does Not Exist Anywhere"); KSR.x.exit(); end if KSR.dispatcher.ds_select_dst(routing_target, 4) then send_reply(404, "No destination"); KSR.x.exit(); end ksr_route_myheader(); ksr_route_relay(); KSR.x.exit(); end /usr/sbin/kamailio[26763]: ERROR: app_lua [app_lua_api.c:713]: app_lua_run_ex(): error from Lua: /etc/kamailio/lua/kamailio-kemi-lua.lua:131: attempt to index field 'dispatcher' (a nil value) kamailio -v version: kamailio 5.0.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 on 20:20:24 Nov 16 2017 with gcc 4.8.5 rpm -aq | grep kam kamailio-mysql-5.0.4-26.el7.centos.x86_64 kamailio-5.0.4-26.el7.centos.x86_64 kamailio-lua-5.0.4-26.el7.centos.x86_64 kamailio-ims-5.0.4-26.el7.centos.x86_64 rpm -ql kamailio-5.0.4 | grep disp /usr/lib64/kamailio/modules/dispatcher.so /usr/share/doc/kamailio/modules/README.dispatcher /usr/share/kamailio/dbtext/kamailio/dispatcher -- Kind Regards *Karsten Horsmann*

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"Federated" PBX solution with GUI to use with Kamailio - Suggestions needed
by George Diamantopoulos 23 Nov '17

23 Nov '17

Hello all! I'm looking at "PBX" platforms to use with kamailio as a load-balancer. This is not a strictly kamailio question, but I thought that if anyone has a suggestion to make, they would be watching this list. I'm investigating a use case with the following requirements: * Kamailio will act as a load-balancer, dispatching initial requests to PBX entities (e.g. Asterisk, Freeswitch) * PBX solution must have a single GUI interface to manage PBX functions (think FreePBX, FusionPBX etc) * Configuration via the GUI must apply to all PBX instances * Some sort of "asterisk-realtime + device state synchronization"-like setup must ensure every PBX box can service any request at any time (for failover) * Solution needn't be multi-tenant oriented, but if it is, it must work fine for single-tenant setups as well without multi-tenant capability getting in the way all the time * Feature list needn't be as full as, say, FreePBX's, but most common functionality must be available (inbound/outbound routing, trunks, queues, hunting to name a few) * Must be open source As a proprietary example, MiRTA PBX seems to check all the boxes above, apart from being open source. Now this "product" also offers high availability for databases and web apps, which is beyond the scope of this investigation. I'm only looking at a solution that provides a federated, synchronized farm of PBX instances. Any input would be greatly appreciated. Thanks! Best regards, George

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Re: [SR-Users] Kamailio IMS + Asterisk PBX
by Amar Tinawi 22 Nov '17

22 Nov '17

My tests results : Unfortunately, no scenario was success, maybe because Asterisk does not support sh interface (Diameter) My test was for presence service and couldn’t get it done. The results of Having asterisk as an AS with kamailio IMS : with sip trunk configured towards S-cscf And whether the user is identified in Asterisk as extension user or not asterisk responded with 404 user not found. Sent from Mail for Windows 10

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debian setup without registration
by Atux Atux 22 Nov '17

22 Nov '17

Hi. i am new to the are of kamailio. i am looking to install in a PC running debian 9 kamailio. that is easy since all i had to do was apt-get install kamailio. I am looking on how to setup the system in a way that everyone sending a register, to be able to register no matter what user name or passwd sends. i do not want a MYSQL server on that server. Could someone give me some hints on how to configure it please? In /etc/kamailio/kamctrlc i have SIP_DOMAIN=192.168.124.5 which is my IP. Everybody from my LAN 192.168.124.0/24 network must be able to send a register with any username and any passwd and be able to register. Also from the outside world, through port forward from my router. Some hints please?

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kamcmd gives segmentation fault (5.0.4)
by José Lopes 22 Nov '17

22 Nov '17

Hello, I invoke kamcmd inside Freeswitch Lua Scripts and right now i am getting segmentation fault any time. This started to happen after I upgraded kamailio from 4.4 to 5.0.4 . I tried to analyse the core dump and it give me the next information. When I execute on the console, the command works. Am I doing something wrong? There is any alternative or better way to communicate with kamailio? Thanks in advance Best regards, Jose Lopes Core was generated by `/usr/sbin/kamcmd -s udp:127.0.0.1:2050 uac.reg_refresh s:29010017'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 591 kamcmd.c: No such file or directory. (gdb) bt #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 #1 0x0000000000403eee in run_binrpc_cmd (s=3, cmd=0x7ffec898a8d0, fmt=0x0) at kamcmd.c:924 #2 0x0000000000404499 in run_cmd (s=3, cmd=0x7ffec898a8d0, format=0x0) at kamcmd.c:1068 #3 0x00000000004073aa in main (argc=5, argv=0x7ffec898d1e8) at kamcmd.c:2114

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Kamailio IMS + Asterisk PBX
by Amar Tinawi 21 Nov '17

21 Nov '17

Dears; Good day ! is it possible to use Asterisk pbx server as an Application server for the kamailio IMS users ? as if i want to correlate an IMS user with an extension number. My though is kamailio ims will handle the registration process and authentication and Asterisk will deal with IMS users as they are PBX users to get all asterisk's benefits (call conference, monitoring, waiting, voice mail ... etc) i'm thinking to integrate Asterisk with S-cscf, and when an IMS user register to the IMS to domain, a trigger will initiate the S-cscf to send notify to asterisk . i didnt find any solution or tutorial for this procedure, and idont know if it is possible to be done. Regards Ammar

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Adding strip and preix attribute to Dispatcher table
by Ashutosh chaubey 21 Nov '17

21 Nov '17

Hi, Is there a way to add prefix and strip in dispatcher table attribute column. I have tried prefix=11,strip=3 but this seems to not work. Any help? Regards, Ashutosh Chaubey

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http_async_client problem with tls_ca_path
by gmele 20 Nov '17

20 Nov '17

Hello, I'm having strange behavior with the http_async_client module and https connections. Our kamailio config use REST interfaces to send push to mobile apps when calls are initiated. We also use a REST interface to register the client in our push database (and then be able to use the info to send a push). Our REST interface use TLS to cypher exchanged data between the SIP registrar and the Pusher service. We have configured the tls connection using modparam: # ------ http async client params ------- modparam("http_async_client", "workers", 16) modparam("http_async_client", "connection_timeout", 20000) # 20 seconds timeout on HTTP requests modparam("http_async_client", "hash_size", 2048) # update this value depending of the load #modparam("http_async_client", "curl_verbose", 1) # curl debug modparam("http_async_client", "tls_ca_path", "/etc/kamailio/ssl/ca") What we have noticed is when there is 1 SIP Register requests triggering and HTTPS request all works fine, but as soon as we have several HTTPS request in //, the HTTPS request fails with curl error 77: curl: (77) Problem with the SSL CA cert (path? access rights?). We tried to put the Root CA Path when building the request: $http_req(all) = $null; # reset the parameters $http_req(method) = "POST"; $http_req(hdr) = "authorization: "+$hdr(authorization); $http_req(hdr) = "Content-Type: application/json"; $http_req(suspend) = 0; # don't suspend the transaction, continue routing script's execution $http_req(body) = "{ 'pushRegistration':{'callId':'" + $ci + "', 'fromUri':'" + $fU + "@" + $fd + "', 'contactLine':'" + $ct + "'}}"; $http_req(tls_ca_path) = "/etc/kamailio/ssl/ca/"; http_async_query("https://pusher:443/rest/push/register", "HTTP_REGISTER_REPLY"); But we have exactly the same behavior. Does the http_async_client module supports muti-thread? Regards Giovanni -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Re: [SR-Users] 183 acc records even if early_media equals to 0
by Daniel-Constantin Mierla 20 Nov '17

20 Nov '17

Hello, can you run list command in gdb to see the code? Because I am not sure what is exactly at that line in your version with the patch. Cheers, Daniel On 09.11.17 18:10, Marco Capetta wrote: > I applied the patch but kamailio crashes with the following error: > > (gdb) bt full > #0 0x00007f8af1e9b9ae in relay_reply (t=t@entry=0x7f8a29c05868, > p_msg=p_msg@entry=0xffffffffffffffff, branch=branch@entry=0, > msg_status=<optimized out>, > cancel_data=cancel_data@entry=0x7ffd92443570, > do_put_on_wait=do_put_on_wait@entry=0) at t_reply.c:1968 > relay = 0 > [...] -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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Dmq usage
by Aidar Kamalov 20 Nov '17

20 Nov '17

Hello, I want to distribute some messages-actions for redis to my kamailio servers. I use this code: For send: dmq_bcast_message("redis", "set early:$fU", "text/plain"); For recieve: request_route { xnotice("ROOT: $rm $rU [$ci]"); if(is_method("KDMQ")) { if($rU == 'redis') { route(REDIS); } else { dmq_handle_message(); } } ...... route[REDIS] { xnotice("REDIS: $rb"); redis_cmd("redis", "$rb", "r"); return; } When dmq_bcast_message starts - other node recieve 7 KDMQ sip packets. And not always it executed. Is it right way to use dmq? Dmq user locations are ok.

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kamailio-wiki dead links to github.com/....master branch
by Karsten Horsmann 20 Nov '17

20 Nov '17

Hello, https://www.kamailio.org/wiki/devel/config-engines has links to https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… for example configs. All the "kemi" stuff is gone. 404 Error. Where i found actuall (kemi / kamailio 5.0.x) samples? -- Kind Regards *Karsten Horsmann*

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no event received for call disconnection
by Aleksandar Sosic 20 Nov '17

20 Nov '17

Hi Everyone, we're trying to implement cgrates in our kamailio nodes. Everything is working fine except for the fact that kamailio is not signaling the call end to cgrates for prepaid users... The versions, logs and confs are: ``` # kamailio -v version: kamailio 5.0.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 5.3.1 ``` ``` # cgr-engine --version CGRateS 0.9.1~rc8 git+15a0793 (2017-10-23T12:15:40+02:00) ``` The kamailio conf regarding our problem is: ``` # Inform CGRateS about CALL_END (stop debit loops, perform accounting if desired in this way) route[CGR_CALL_END] { if $sht(cgrconn=>cgr) == $null { xlog("Charging controller unreachable"); exit; } $var(callDur) = $TS - $dlg(start_ts); evapi_relay("{\"event\":\"CGR_CALL_END\", \"callid\":\"$dlg(callid)\", \"from_tag\":\"$dlg(from_tag)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_answertime\":\"$dlg(start_ts)\", \"cgr_duration\":\"$var(callDur)\", \"cgr_supplier\":\"$dlg_var(cgrSupplier)\", \"cgr_disconnectcause\":\"$T_reply_code\"}"); } ``` The kamailio logs: ``` 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} ``` Thanks for your help, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

2 13

kamailio 4.4 - jansson_get_field parse error actions
by Jonathan Hunter 17 Nov '17

17 Nov '17

Hi Guys, I am looking to send calls to a default route should we have issues when connecting to our API which provides routing instructions. On scenario is handling an invalid JSON response , which would result in a parse error. Is it possible to capture when there is a failed to parse error when dealing with JSON responses to an http query in kamailio? In particular when using the jansson_get_field, I am looking to default to a route should their be a parse error, so I am forcing an erro, I get this in the logs, but can I grab to act on it? Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' On receipt of this I would then like to hit a default route. Many thanks Jon

1 1

SUBSCRIBE NAT Keep-Alive
by Asgaroth 17 Nov '17

17 Nov '17

Hi All, I'd like to ask the guru's out there their opinion on how to solve the nat problem for presence subscriptions. We have an issue with subscriptions that get timed out and the notify clears the subscription out of the active_watchers table due to the notify timing out, I think this is expected and the right way to do it. However, how do others out there ensure that their subscription nat states are kept alive? For registrations we have nathelper, but, what is the sanest way of doing the same for subscriptions? I was originally thinking of sending notifications through our registrar/lookup server, but, I'm not sure if that is the correct way to do it. Any thoughts, tips, tricks would be much appreciated. Cheers

3 3

Kamailio questions
by Wilkins, Steve 17 Nov '17

17 Nov '17

Hello, I am new to Kamailio and to the Kamailio Mailing list and hope I am using it appropriately. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. For the sake of the argument, my server name is myServer, and it's IP Address is 172.1.1.13; Asterisk is on a different server. Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 myServer /usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.1.1.13:59794 failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work; I can make calls and use sip tools which pass requests through the Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what at such a high port. Question: The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Note: Even with no rtpproxy, I get the same error mentioned above in the Issue section. Thank you,

2 4

dr_load_routing_info(): failed to add rule id 4 -> skipping
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i experiment with the drouting module with mysql-db. I try to route with "alphanumeric" stuff - like 'tetris' in sip://tetris@mytest.example https://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp31352844 The DB-Field is an 64 string like route-id. In dr_rules.route-id i can use "named-routes" if i leave the prefix empty or i fill in only numbers. With strings i got this error: ERROR: drouting [dr_load.c:555]: dr_load_routing_info(): failed to add rule id 4 -> skipping SQL Result for rule-id 4: # dr_rules # ruleid, groupid, prefix, timerec, priority, routeid, gwlist, description '4', '9000', 'tetris', '', '0', 'tetris', '4', 'testroute' Is drouting only for "number-routing"? -- Kind Regards *Karsten Horsmann*

3 6

best practice to dynamic route block based on $rd - Domain in R-URI
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i am now at the point that one kamailio is not enough :). So i decide to set another kamailio internal as sip-router based on drouting. My config is straight forward the example config with additional drouting stuff. Whats the best practice to route based on $rd with kamailio? I dont want to end with an bunch of if then else. Should be come from database. Pseudo Code: if ($rd == old.sbc.domain) { route[oldstuff]; } else { route[newstuff]; } Is there an module that i can use? -- Kind Regards *Karsten Horsmann*

1 1

How to use $var as arguments to module functions
by Duarte Rocha 16 Nov '17

16 Nov '17

Greetings, Is there a way to use the NATHELPER functions with a variable insted of a string with the value ? For example, I can do "fix_nated_sdp("11");" but Kamailio throws an error if i use "fix_nated_sdp("$(var(nat_fix)");" or any other variations. How should i proceed to be able to do this ? Thanks

2 1

play music while processing a http request
by 赵国杰 16 Nov '17

16 Nov '17

Hello guys, I have the following senario: ua1 calls ua2. when the INVITE reaches the kamailio, before foward to ua2, kamailio has to do a http request using “http_async_query”. However, the http request can take a really long time. I want the kamailio start playing music when the http request is sent and stop the music when the http response is received. My config file is like this: request_route { ... t_newtran(); http_async_query("url here", "HTTP_REPLY"); } route[HTTP_REPLY] { ... # forward the invite to ua2 } how do i complete the config? Thanks

2 3

Dmq_usrloc
by sofos koko 14 Nov '17

14 Nov '17

Hi ...i would like to connect two kamailio in diffwrent network so i can get call from one to another is any kamailio.cfg example to help me do it!

1 0

topos and topos_redis : error log -> ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store
by Abdoul Osséni 14 Nov '17

14 Nov '17

hello, After activated redis backend for topos module, I have some error logs. Can you help me to understand these logs? # kamailio -V version: kamailio 5.1.0-rc0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 on 14:18:08 Nov 14 2017 with gcc 4.9.2 Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Abdoul OSSENI

1 0

AVPOPS: is_avp_set/avp_check "name" parameter as variable.
by Asgaroth 14 Nov '17

14 Nov '17

Hi All, I was having a read of the AVP module documentation and I see mentioned that in future releases of kamailio that the AVP module may become deprecated in favour of using AVPOPS module. To this end, I started looking to switch some of AVP functions over to AVPOPS functions when I cam across an issue where I cannot have the name parameter of the avp to check a variable. I have some attributes defined for a domain which I load up using lookup_domain, one of these attributes is a dispatcher setid based on the request method being processed. So, with AVP I can do something like the following to see if it is set: if ( attr_exists( $_s(dm_ds_setid_$rm) ) ) The above works as expected, but if I try using something equivelent using avpops functions, something like: if ( is_avp_set("$avp(s:$_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(s:dm_ds_setid_$rm)") ) or if ( is_avp_set("$avp($_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(dm_ds_setid_$rm)") ) Then it fails the check even though the avp is set correctly, there are no errors in the logs so it all appears to be okay. Likewise, when checking with the following AVP function, it works: if ( attr_equals("$_s(dm_ds_setid_$rm)", "100") ) But, if I try with AVPOPS function, it fails (I'm guessing its bacause the avp cannot be found) if ( avp_check("$avp(s:$_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(s:dm_ds_setid_$rm)", "100") ) or if ( avp_check("$avp($_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(dm_ds_setid_$rm", "100") ) Are the AVPOPS functions able to take nested variables like the above? Is this a bug or am I using them incorrectly? I'm currently trying this out on kamailio v5.0.4. Any thoughts on the issue would be appreciated. Thanks

2 2

max_contacts does not work as expected
by Gholamreza Sabery 14 Nov '17

14 Nov '17

I have a Kamailio server that has: modparam("registrar", "max_contacts", 2) But recently I noticed some users have more than two records in the location table. For example one of the users has three different records with nearly the same information. The records are as follows. Is this behavior normal? (I am using Kamailio 4XX). *************************** 1. row *************************** id: 171296745 ruid: uloc-2c69-59db6c73-7ca-132d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: XpCnmaAhRF cseq: 43 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 14 *************************** 2. row *************************** id: 171296746 ruid: uloc-2c69-59db6c73-7f2-eb5d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: BzMKuyJN8C cseq: 50 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 23 *************************** 3. row *************************** id: 171296747 ruid: uloc-2c69-59db6c73-80f-541d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: 3p9P4PgZSv cseq: 51 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 15

1 0

Load Gateway according hostname
by Ashutosh chaubey 13 Nov '17

13 Nov '17

Hi, I am trying to find a way to load a specific gateway but instead of lcr_id i want it load gateways according to the hostname. I am passing Hostname as a header to kamailio so its easy to access that but unabale to load gateways according to it. Any help? Regards, Ashutosh Chaubey

1 0

using bcrypt passwd hashing
by Walter Martín Villalba 13 Nov '17

13 Nov '17

Hello, I did some searches online and talked to some colleagues and it seems Kamailio only supports the traditional HTTP digest authentication, which uses MD5. I would like to know if any of you has been successful in using bcrypt/scrypt/pbkdf2 passwd hashing, instead of MD5, which has been deemed as obsolete and insecure a long time ago. Perhaps you've written your own auth module, or just modified the config script to call some other credential checking routine using a custom python/perl script (I'm thinking of doing the latter, of nothing better is available). If any of you have done something like this, using bcrypt or any other current and secure hashing algorithm, I would appreciate some guidance. If you haven't, aren't you concerned about storing MD5 password hashes in your database? Note: if I can't find a good answer using this list, I will try the developer's list next. Thanks in advance, Martín.

4 5

Kamailio issue
by bazou2 bazou2 13 Nov '17

13 Nov '17

Hello, I have a kamailio version 4.4 with an Asterisk 13 ... and i need to make a call from a sip phone app.(i have a user subscribed on the kamailio to be used on the sip phone app). The weird thing that first time, the call works (I can see the sip messages working fine between the app kamailio and asterisk) but when i try again it doesn't work. SIP messages show that the kamailio and Asterisk keep send trying request between them and the loop of these kind of messages doesn't end.. I need to know if i am missing something in the configuration file of the kamailio or what? Regards Sent from Outlook<http://aka.ms/weboutlook>

2 2

Development open in master branch (to be v5.2.x)
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, git branch 5.1 was just created (to host the release series v5.1.x), therefore new features can now be pushed again in master branch. They will be part of the next future release, likely to be numbered 5.2.x. Any fixes that affect existing code in branches 5.1 or older version have to be backported - push first to master and then cherry pick -- see the contributing guidelines at: - https://www.kamailio.org/wiki/devel/git-commit-guidelines#backporting_commi… Many thanks to all contributors so far! Testing of branch 5.1 and giving feedback for it is very appreciated! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Branch 5.1 created
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, the GIT branch 5.1 has just been created, it will host the release series 5.1.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 5.1 origin/5.1 Hopefully in two-three weeks time frame the full release of 5.1.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 5.1. No new features can get in branch 5.1. Enhancements to documentation or helping tools, as well as kemi exports are still allowed. If you are not sure about doing or not a backport, ask on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

GeoIP2 module deb package
by Joel Serrano 12 Nov '17

12 Nov '17

Hi, I can't seem to find the geoip2 module for debian stretch (installing from packages, kamailio 5.0 branch). Is it not available in the repository? Thanks, Joel.

2 4

Kamilio for IMS and VoLTE Setup Questions
by Jack Folgolburg 11 Nov '17

11 Nov '17

Hello, I saw the Kamailio World 2017 presentation "Kamilio for IMS and VoLTE" (https://www.youtube.com/watch?v=Wdhcdv5JVPw). I would like to install the servers for evaluation and testing. The slides say to use Ubuntu 16.04, but there are no xenial packages at the ng-voice repository. Is it okay to install on Ubuntu 14.04, since there are trusty packages? I have done this and got through all the configuration prompts. After installation, is there any way to confirm that the site is operational? How to bring up the management console (i.e. what is local URL)? In the presentation, it looks like a commercial site at ng-voice is accessed. I understand that there is a REST API. Is this API documented anywhere? Maybe, the most important question is, after installation, what is the fastest way to add two users and test access from IMS clients? Thank you, James

1 0

183 acc records even if early_media equals to 0
by Marco Capetta 09 Nov '17

09 Nov '17

Dear All, I'm facing a strange problem with the call accounting module: even if in my configuration I have the parameter: modparam("acc", "early_media", 0) I can find some ACC records with sip_code 180 or 183. I investigated those cases and this issue seems to happen when an endpoint sends an 183 message immediately followed by a 200 OK. The issue seems started from kamailio version 4.4.1, but I cannot be really sure of this. Have you ever had a similar problem? Do you have any suggestions on how to solve it? Thanks Marco

2 11

t_relay_to_udp - 500 server error occurred (6/SL)
by Donat Zenichev 09 Nov '17

09 Nov '17

Hi all. I have a problem with t_relay_to_udp function. My topology: UAC -> tcp -> kamailio <-> udp <-> asterisk Kamailio processes requests from clients over TCP and relays requests to asterisk over UDP. Kamailio gets INVITE from client, relays it to asterisk, then he sends 100 trying to user client and abruptly sends 500 internal error to client (after 100 trying). But the main interesting thing is, that asterisk answers on INVITE and dialog continues between kamailio and asterisk (without user agent client) - it reaches the destination then, without originator : ) How error on kamailio looks like: Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: tm [t_fwd.c:1716]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (6/SL) The same problem is, when I change transport protocol between kamailio and asterisk to TCP, using the t_relay_to_tcp function. When I use t_relay function, it works fine, nothing happens that can drop dialogs between kamailio and clients. Relaying code part: if ($var(to_asterisk)=="1") { xlog("L_NOTICE","Relaying to asterisk - change socket to udp:10.1.1.1:5070 \n"); force_send_socket(udp:10.1.1.1:5070); if (!t_relay_to_udp()) { sl_reply_error(); } } I need to use t_relay_to_udp with force_send_socket, because kamailio changes the source port (source socket) every time he generate new INVITEs to asterisk and this is irrelevant. Dump between kamailio and user agent client: INVITE sip:1111111111example@kam1.domain.com:5070;transport=TCP SIP/2.0 Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z- Max-Forwards: 70 Contact: <sip:debug.device-example@uac.domain.example:5065;transport=TCP> To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="debug.device-example",realm=" kam1.domain.com ",nonce="WgQGlVoEBWnhH3CF6DE77uKxKJx0tT4e",uri="sip:1111111111example@kam1.domain.com:5070 ;transport=TCP",response="4753aa849715d26d0dde1b3c2314c843",a orithm=MD5 Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 v=0 o=Z 0 0 IN IP4 uac.domain.example s=Z c=IN IP4 uac.domain.example t=0 0 m=audio 8000 RTP/AVP 8 3 110 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: MS Lync Content-Length: 0 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL) Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070 ;transport=TCP>;tag=9a78ee8ed9a3d4ba441c41d83ba2e904.4113 From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: example Content-Length: 0 Thanks in advance for hints. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-800 http://wnet.ua

1 0

Topoh module configuration for SIP proxy and SIP registrar
by Christian Conus 08 Nov '17

08 Nov '17

Hello, I am trying to configure the topoh module for kamailio 5.0.2. I read the documentation but some points are still not clear to me. The topology of my network is as follow: - Two SIP proxies - Two SIP registrars - Only one SIP proxy is running (the other take over automatically in case the first fails) - Both registrars are running (proxy sends requests to both registrars using round robin algorithm) Text mode graph of the topology (hope it helps understand): I tried to enable topoh on SIP proxy only, on both proxy and registrars, with same mask_key everywhere and with different mask_key between proxy and registrars but everytime I could still see parts of the SIP header which was not obfuscated. My questions are: - Is it necessary to enable topoh module on each SIP proxy and SIP registrar ? - The mask_key should be the same on each server or it should be different between proxy and registrars? - the mask address should be the same on each server or it should be different between proxy and registrars ? Thanks in advance for your help, Christian -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 10

evapi_async_relay() failed
by Abdul Basit 07 Nov '17

07 Nov '17

Hi list, I am configuring CGRateS with kamailio 4.4 using deb packages on debian 8. I am facing a issue while processing event_route[dialog:start] for answering call INVITE at evapi_async_relay("{\"event\":\"CGR_CALL_START\", with following error /usr/sbin/kamailio[13887]: ERROR: evapi [evapi_mod.c:288]: w_evapi_async_relay(): failed to suspend request processing However, after discussion with Dan on IRC i replaced function evapi_async_relay() with evapi_relay() resolved the issue. Whats causing this error? Can anyone suggest some way out for keep using evapi_async_relay? -- regards, abdul basit

2 1

Kamailio SIP-I
by Saleem Manasfi 07 Nov '17

07 Nov '17

Dears, Please if you can let us know how does Kamailio reads the SIP-I body (ISUP)? Is there any document or link that may help?

2 1

Re: [SR-Users] Regarding Opensource DID Routing Solution using Kamailio
by surendra tiwari 06 Nov '17

06 Nov '17

Hi All, I have used the kamailio and redis to create the DID routing solution.please find below in github.this repository is created to help the person who want to setup there own DID routing solution with following feature: 1. Inbound Termination with Carrier IP validation. 2. Carrier LCR for DID/TFN to PSTN forwarding. 3. Inbound Abuse Block. 4. CDR in MongoDB. 5. IPTables Block for Sip-Scanners. 6. Integration with RtpEngine. 7. RedisDB for quick DB Access. https://github.com/surendratiwari3/KamInboundSIP On Mon, Nov 6, 2017 at 9:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > your email was addressed to kamailio-announce mailing list (I took your > message from its bounces filter), but that list is only for announcement > issued by Kamailio project (major news about our releases, security > incidents, events, etc...). > > News from community members are welcome and can be sent to: > > - sr-users(a)lists.kamailio.org for non-commercial announcements > > - business(a)lists.kamailio.org for announcements related to commercial > offerings/products (but here you can also send non-commercial announcements) > > Your announcement is not commercial and can be sent to sr-users. Once it > will be there, I will also make a news article on kamailio.org website. > > I didn't have the time yet to check in detail what you put together, but > sounds interesting. > > Thanks for sharing, > Daniel > > (cc-ed Fred as he is moderator for kamailio announce mailing list) > > On 01.11.17 18:46, surendra tiwari wrote: > > Hi All, > > I have used the kamailio and redis to create the DID routing > solution.please find below in github.this repository is created to help the > person who want to setup there own DID routing solution with following > feature: > > > 1. Inbound Termination with Carrier IP validation. > 2. Carrier LCR for DID/TFN to PSTN forwarding. > 3. Inbound Abuse Block. > 4. CDR in MongoDB. > 5. IPTables Block for Sip-Scanners. > 6. Integration with RtpEngine. > 7. RedisDB for quick DB Access. > > > https://github.com/surendratiwari3/KamInboundSIP > > -- > With Regards, > Surendra Tiwari > VoIP Application Developer, > +91-9967609476 <+91%2099676%2009476> > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > -- With Regards, Surendra Tiwari VoIP Application Developer, Plivo Comm +91-9967609476

1 0

Sipcapture HEP Values
by Soltanici Ilie 06 Nov '17

06 Nov '17

Hi all, How can i get the Destination IP from HEP packet? I tried $hep(dst _ip) like in documentation (https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#sipcapture_mo…) - but is not working. (pvar "hep" has an invalid name param [dst_ip]). Thank You.

1 0

Invalid JSON when dumping htable with JSONRPC-S
by Marrold 03 Nov '17

03 Nov '17

Hi All, I am currently trying to retrieve the contents of a htable using the JSONRPC-S module over HTTP, however I have observed if a slot contains multiple values, the JSON returned has duplicate keys which could be considered invalid. Here are the relevant lines from kamailio.cfg - event_route[xhttp:request] { if ($hu =~ "^/carrier") { jsonrpc_exec('{"jsonrpc": "2.0", "method": "htable.dump", "params" : {"htable": "carrier"}, "id": 1}'); xhttp_reply("200", "OK", "text/html", "$jsonrpl(body)"); } } And the output, showing a slot with two items- { "entry": 5, "size": 2, "slot": { "item": { "name": "carrierA", "value": "90", "type": "str" }, "item": { "name": "carrierB", "value": "190", "type": "str" } } } And the same from `kamcmd -s unix:/tmp/kamailio_ctl htable.dump carrier` - { entry: 5 size: 2 slot: { item: { name: carrierA value: 90 type: str } item: { name: carrierB value: 190 type: str } } } Would it be possible to change it to something like the following - { "entry": 5, "size": 2, "slot": [ { "name": "carrierA", "value": "90", "type": "str" }, { "name": "carrierB", "value": "190", "type": "str" } ] } Any other suggestions or work arounds would be appreciated, Thanks Matthew

2 1

Contacts in 302
by Diego Nadares 03 Nov '17

03 Nov '17

Hi Guys, I'm generating a 302 reply from kamailio. In this 302 I append new branches with new contacts. if($var(routing)=~"redirect"){ jansson_get("contacts_len", "$var(evmsg)", "$var(contacts_len)"); xlog("L_INFO", "Contacts len $var(contacts_len)"); $var(i) = 0; while ($var(i) < $var(contacts_len)){ jansson_get("contacts[$var(i)]", "$var(evmsg)", "$var(contact)"); append_branch($var(contact), "0.5"); $var(i) = $var(i) + 1; } send_reply("302", "Moved Temporarily"); exit; } The problem i'm facing is that I can't delete the original contact *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>* The resulting conctact: Contact: *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>*, <sip:1111@188.111.111.112>;q=0.5, <sip:1111@188.111.111.112>;q=0.5 I all ready tried with remove_hf('Contact') and adding the new ones after that but it doesn't worke either. The problem is that one gw takes the first contact over and over again and never the other two. Thanks in advance. Diego.

2 4

Multiple IPs
by Oz Mortimer 03 Nov '17

03 Nov '17

Hi all, I have Kamailio set as an SBC with freeswitch behind - that all works fine. Freeswitch can send calls out VIA Kamailio to external IP address’, but if I try sending a call to another IP hosted by Kamailio the seems to go nowhere… Supplier -> Kamailio -> Customer1 -> Kamailio -> Customer2 The Kamailio server has multiple assigned IPs, which are used to multi-tennant - but I cant for the like of me to get Tennant to tenant calls to work (via freeswitch) Any pointers?

2 13

onreply_route block not processing replies
by José Seabra 03 Nov '17

03 Nov '17

Hi there, I'm facing an issue regarding with replies coming to kamailio that aren't processes inside of onreply_route block. Anyone here can help me understand why these replies (1XX) aren't entering on onreply_route bock? is there any situation already identified on Kamailio that can originate this behavior? The Kamailio version used is 5.0.1. Thank you Regards -- José Seabra

2 3

Branching Kamailio v5.1
by Daniel-Constantin Mierla 03 Nov '17

03 Nov '17

Hello, I am planning to create the got branch 5.1 (to be used for v5.1.x series) by end of next week (likely on Friday, Nov 10, 2017). After that, the master will be open for new features and fixes will need to be backported to 5.1. Should anyone want a different schedule, reply to the mailing lists and propose your plans. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

WebRTC error code 488 in iOS application
by Denys Pozniak 03 Nov '17

03 Nov '17

Hello! Probably this case is not connected Kamailio directly, but it can be somebody will point me in the correct direction :) We have VoIP solution: FreeSWITCH -> Kamailio + Rtpengine -> WebRTC Chrome. Everything works Ok except iOS application based on React Native WebRTC ( https://github.com/oney/react-native-webrtc) + JsSIP. It answers error 488 on incoming INVITE (wSDP): INVITE SDP to iOS: *......* *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* iOS console log: *11:55:23.670 browser.js:133 JsSIP:WebSocketInterface send() +10ms* *11:55:23.670 browser.js:133 JsSIP:RTCSession session progress +5ms* *11:55:23.671 browser.js:133 JsSIP:RTCSession emit "progress" +1ms* *11:55:23.671 webrtc.js:108 WebRTC.on progress: local* *11:55:23.679 browser.js:133 JsSIP:RTCSession emit "peerconnection" +8ms* *11:55:23.680 webrtc.js:89 WebRTC.on peerconnection* *11:55:23.680 browser.js:133 JsSIP:RTCSession emit "sdp" +1ms* *11:55:23.689 browser.js:133 JsSIP:Transport send() +20ms**11:55:23.689 browser.js:133 JsSIP:Transport sending message:* *SIP/2.0 488 Not Acceptable Here* *11:55:23.701 debug.js:115 JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:Error: Failed to set remote offer sdp: Failed to create channels.* I found similar issue on thier git https://github.com/oney/react-native-webrtc/issues/293 and main solution is to struct SDP in way like: *a=group:BUNDLE audiom=audio PORT RTP/SAVPF N M Ka=mid:audio* How to modify SDP on Kamailio side (after Rtpengine modification) like below? *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *a=group:BUNDLE audio* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=mid:audio* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* BR, Denys

1 1

User Registration with location awareness
by Daisy Stevens 03 Nov '17

03 Nov '17

Hi Guys Please forgive me if I seem to asking stupid questions but I am a complete dummy at the moment regarding this software. I would love to be able to set up two kamailio servers with the idea of expansion in the future to handle SIP registrations, however having a two server model how would I get the calls to route to another user that is registered on the other kamailio box? Kind Regards Daisy -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

2 1

msilo Content-Type for picture and video?
by sscc 03 Nov '17

03 Nov '17

Hello kamailio folks i am successfully storing and dumping messages for off line users using msilo content type text/plain. i want to store off-line messages for pictures and videos. my particular queries are 1) is msilo store picture and videos? (i have observed blob in silo database table which is used for pictures. 2) in case it does what will be declared in content type hdr parameter? any help in this regard will highly be appreciated. regards, sscc -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Re: [SR-Users] Reading the P-Assserted-identity header
by Gertjan Wolzak 02 Nov '17

02 Nov '17

Alex and Daniel, Thank you for the feedback, the tip from Daniel helped me pinpoint the problem. And the the one from Alex showed that. The problem was between keyboard and chair, I should have seen that the SBC removed the PAI. Once I changed that, the PAI arrived at Kamailio and manipulation was a breeze. Thanks guys.

1 0

Tcp socket error with active/active kamailio schema
by Donat Zenichev 02 Nov '17

02 Nov '17

Hi guys. I'm trying to realize active/active schema with kamailio. Topology is as following: active/active kamailio -> routing server (asterisk) Kamailio servers work as registrar/location servers and sbc. Asterisk server performs routing functions for calls. We use sip / sips / ws / wss for signaling. Schema built on aws platform and both kamailio have virtual ip address. So when first kamailio crashes, second one assigns ip address of first kamailio. And vice-versa. Both kamailio listens on both shared addresses all the time (with help of sysctl non lolcal bind function). Problem occurs when first kamailio goes down and second kamailio gets ip address of first one. Asterisk tries to send in-dialog requests (for dialogs that were established on first kamailio), request goes to second kamailio and asterisk gets a socket error. And this is proper behaviour I think, because first kamailio established tcp connection with asterisk from certain port, that is not listened on second kamailio. So that, I think that the solution is to use force_send_socket function, but I'm not confident about it. If anyone has ideas, please send your advices. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Kamailio not calling each time
by hanoulia salameh 02 Nov '17

02 Nov '17

Hello, I have a problem with kamailio (v4.4), when i try to call from a sip phone through kamailio, the call is interrupted and hang up automatically but after 4s it continues to retry (I can see in the log file Session Progress) and goes through the asterisk (certified/13.13-cert4) and establishes a normal call (It rings at the client side) while it's still disconnected from the sip phone side But when i try again it works fine sometimes. Am I missing something in my configuration file? What can I show in order to help me? Regards

2 2

Kamailio Realtime Integration with Asterisk
by Floimair Florian 02 Nov '17

02 Nov '17

Hi! I found a few tutorials from Daniel on this topic , the latest being https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astd… . Has anyone ever tried this with a more recent Asterisk version (13 or newer) and using chan_pjsip instead of chan_sip? I'd be willing to update the tutorial once I get it running but I would like to avoid going through obstacles that were already handled ;-) My desired setup is 2 Asterisk boxes (which share configuration, useres and registrations via DB) behind a Kamailio as Load-balancer (got that working already). With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 f.floimair(a)commend.com<mailto:f.floimair@commend.com> http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg

1 0

Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 02 Nov '17

02 Nov '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

4 11

Reading the P-Assserted-identity header
by Gertjan Wolzak 01 Nov '17

01 Nov '17

Hello Kamailions, I receive anonymous calls where the PAI has a prefix which I need to strip before forwarding the call to a proxy. I am trying to read the the header with $hdr(P-Asserted-Identity) and I tried the $ai pseudovariable, but I get an empty reply back every time. xlog("PAI is $ai"); results in ; "Nov 1 16:19:55 proxy04 /usr/local/sbin/kamailio[14814]: ERROR: <script>: PAI is <null>" Same for xlog("PAI is $hdr(P-Asserted-Identity). This works for the Privacy header but not the P-Asserted-Identity header. Hope someone can tell me what I am doing wrong.... Rgds, Gertjan

3 2

Record-Route IP Value
by Soltanici Ilie 31 Oct '17

31 Oct '17

Hi All, Having the following Header: Record-Route: <sip:192.168.1.1;lr;did=637.07c7c2d7> How can i extract only IP value from this header? Thank You.

2 3

module version mismatch for /usr/local/lib64/kamailio/modules/kex.so
by Abdul Basit 31 Oct '17

31 Oct '17

Hi, My kamailio installation with 5.0.3 was working fine. I update kamailio source to 5.0.4 and make all; make install I started receiving error below: /etc/init.d/kamailio start [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(7426) ERROR: <core> [core/sr_module.c:431]: version_control(): module version mismatch for /usr/local/lib64/kamailio/modules/kex.so; core: kamailio 5.0.3 (x86_64/lin[FAILmodule: kamailio 5.0.4 (x86_64/linux) ... failed! failed! kamailio.cfg is same as earlier. I removed kamailio sources and modules folder /usr/local/lib64/kamailio/modules/ I also dropped old kamailio postgres database and created new one after fresh source install of version 5.0.4 from git. # git clone https://github.com/kamailio/kamailio kamailio # cd kamailio # git checkout -b 5.0 origin/5.0 But this error persists. Whats the resolution? Is this a bug? -- regards, abdul basit

2 2

Source comparisons by DNS hostname
by Alex Balashov 31 Oct '17

31 Oct '17

Hi, Is there a reasonable way, in Kamailio, to determine whether the source of a given SIP message is equivalent to the IP address to which a given DNS hostname resolves? i.e. if(src_ip == 'fqdn.domain.xyz') but not src_ip, obviously. I can't find any core keywords that would seem to fit the bill for a composite comparison. Any other options? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Radius accounting and cdr's
by Jan Hazenberg 31 Oct '17

31 Oct '17

Hi all, I have a question regarding radius accounting. I would like to have a config where i store local cdr records (and not the ACC records) and send accounting to radius as well. I have the following config: # ----- acc params ----- /* General settings */ modparam("acc", "log_level", 0) modparam("acc", "log_facility", "LOG_LOCAL2") /* Enable cdr's */ modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_LOCAL2") modparam("acc", "cdr_on_failed", 1) /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) /* Accounting triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACCMISSED) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") This works but stores the raw accounting records as well. If i disable the acc log_flag it stops sending radius accounting records (while the acc_radius log_flag is still enabled). Is it possible to send only radius accounting records and disable the local logging of those records, but leave the cdr logging enabled? Or is there a option where i can set only the cdr_log_level so accounting records will only show up when i increase the debug level? Another option is to use a seperate log facility for accounting and cdr's, but it would be nice if it is possible to disable the logging of accounting records and only keep the cdr's. I'm running kamailio 5.0.3. Thanks, Jan

2 2

Re: [SR-Users] Kamailio for Unified Communication
by Angel Elena 31 Oct '17

31 Oct '17

kamailio is a "proxy" server.... this features may be added by a media-server... freeswitch, asterisk, seems. You can put kamailio as "sip proxy" and you can add these services behind kamailio proxy -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Zarafa Webapp. -----Mensaje original----- De: Jayanth Acharya <jayachar88(a)gmail.com> Enviado: Mar 31-10-2017 10:40 Asunto: [SR-Users] Kamailio for Unified Communication Para: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>; > Hi, > > Can Kamailio be used to provide following broad-type of services ? > > 1. Unified Communication services > 2. Contact Center services > 3. Instant Messaging / Chat services > 4. Voicemail service > 5. Presence aware communication services > 6. Desktop sharing service > > For many/some of the above, I'd expect back2back user-agent capabilities to be > mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are > there some modules that may be used for this purpose ? > > thanks, > Jay > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > >

1 0

Kamailio for Unified Communication
by Jayanth Acharya 31 Oct '17

31 Oct '17

Hi, Can Kamailio be used to provide following broad-type of services ? 1. Unified Communication services 2. Contact Center services 3. Instant Messaging / Chat services 4. Voicemail service 5. Presence aware communication services 6. Desktop sharing service For many/some of the above, I'd expect back2back user-agent capabilities to be mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are there some modules that may be used for this purpose ? thanks, Jay

1 0

Reinvite/ACK race
by Alex Balashov 30 Oct '17

30 Oct '17

Hi, I've got a scenario like so: UA A -----> Kamailio P ----> UA B 1. UA A initiates call through Kamailio P; 2. Dialog is established and confirmed, with Record-Route; 3. UA B sends reinvite #1 through P to A; 4. UA B sends 2xx reply; 5. UA B sends end-to-end ACK for reinvite #1 and almost simultaneously sends reinvite #2. The temporal delta is between reinvite #2 and ACK for reinvite #1 on the wire is 3 ms. So, the result — for all kinds of stochastic processing and userspace scheduling type reasons — is that the reinvite is forwarded first, before the ACK. That leads to a 500 / 491 scenario UA A. The cause, as we all know, is that Kamailio's worker threads are loosely coupled, and incoming UDP datagrams are distributed directly by the kernel. Is there any general guidance on what to do with these scenarios? I looked at RFC 5407 § 3.1.4, which appears to describe a similar, but not identical scenario involving an initial INVITE and subsequent reinvite. As far as I can tell, the recommendation in that standard is "space the messaging out more in time". Switching to TCP would presumably help, since any given flow would involve a single connection to a single worker thread and the transport would guarantee ordering. However, that's not really feasible in this implementation for a host of reasons. I know Kamailio has some config locking primitives, but I am extremely wary of complex synchronisation on dialog-identifying attributes, particularly if there is a possibility that such locks could stall a worker perpetually. Any other thoughts welcome! Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

unexpected behavior of save from registrar module
by Vasiliy Ganchev 30 Oct '17

30 Oct '17

Hi, community! I have in my Kamailio script following block (while REGISTER processing, after auth succeed): ..... /* Save to usrloc */ lock("LOCATION_LOCK"); if (!save("location")) { unlock("LOCATION_LOCK"); sl_reply_error(); exit(); } unlock("LOCATION_LOCK"); ...... According to module documentation - if "save" - failed (for any reason) - I expected the control to be returned to the script with error code "-1" But, there are cases - when this does not happen - which lead to deadlock for workers processing REGISTERs (I never unlock("LOCATION_LOCK")): - module answers with internal error - and return 0 from "save" function: https://github.com/kamailio/kamailio/blob/master/src/modules/registrar/save… - line 1013 My use case happened due to malfunctioning device that in some moment of time: - send REGISTER update - kamailio answers 200OK - device retransmit the same REGISTER (same CSeq) - during 20 seconds (hardcoded value in registrar module) - kamailio answers 200OK (and everything is ok) - if retransmission continue more than 20s - kamailio answered 400 Bad request, and did not return control to script (as a result I had no chance to unlock("LOCATION_LOCK") from code above) Tested it on kamailio 4.2.5 and 5.0.3. as for me, possible solutions are: - return -1 instead of 0 from "save" by default - if there are some reasons not to do this - we should change documentation, and explicitly mention that there are cases - when script execution is stopped. thanks in advance! cheers -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

3 5

sanity passes invalid uri
by Juha Heinanen 30 Oct '17

30 Oct '17

I noticed that K 5.0 sanity() test passes uri that contains ` (back quote) character. Config: modparam("sanity", "default_checks", 1024) /* URI checks */ modparam("sanity", "uri_checks", 3) /* RURI, From */ xlog("L_INFO", "Checking $ru\n"); if (!sanity_check()) xlog("L_INFO", "Check failed\n"); else xlog("L_INFO", "Check passed\n"); Syslog: Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Checking sip:jh@te`st.fi Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Check passed According to RFC3261: hostname = *( domainlabel "." ) toplabel [ "." ] domainlabel = alphanum / alphanum *( alphanum / "-" ) alphanum Is this a bug or am I missing something? -- Juha

2 2

Remove all cpecial characters (\n,\t etc)from xml body
by Yuriy Gorlichenko 30 Oct '17

30 Oct '17

Hi. I building external presentity service for web based phones Main trouble that i have at the xml body of PUBLISH message \n \t characters and because of it can't set body as string to the redis So I trying to find a way to remove special characters form the xml body for get oneline string instead of multiline I use native .cfg file for now and, unfortunattely, can not use lua or phyton I tried $var(body)=$(rb{s.rm,$var(RemoveThisStaff)}) where $var(RemoveThisStaff) = "\n" and "\t" but it was unsuccesfull. Maybe someone will help to kno how to do it.

2 2

Issue with sipcapture
by Nathan Leyton 28 Oct '17

28 Oct '17

Hi There I am new to Kamailio and have just installed and got it up and running as a sipcapture node. It is working well and logging all the hep sip traffic in my homer database. I have an issue with rtcp extracted to the msg column is not json. It is some kind of weird encoded string (not base64), this is not readable by homer in any way. When I use nonsip_hook I am able to extract the json from the hep and put it in the msg field. However when I return 1 from the hep route on sip to go back to main route I get parsing errors like so on all the sip messages:- Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: sipcapture [sipcapture.c:2711]: nosip_hep_msg(): couldn't parse sip message Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/receive.c:186]: receive_msg(): core parsing of SIP message failed (178.78.120.2:44389/1) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [(null)] Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: WARNING: <core> [core/receive.c:193]: receive_msg(): parsing relevant headers failed Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/msg_parser.c:675]: parse_msg(): ERROR: parse_msg: message=<HEP3#002,> It never actually enters the main route. With nonsip_hook off, I enter the main route and all my sip messages are there in homer, just missing rtcp as the data is a weird encoded string. Hope someone can help Thanks Nathan -- Nathan Leyton / Chief Technology Officer nathan(a)callhandling.co.uk / 07482 917 412 Call Handling Ltd Office: 0333 321 0888 / Fax: 0333 321 0877 Bakerloo Chambers304 Edgware Road, London, W2 1DY www.callhandling.co.uk [image: htmlsig.com] <https://htmlsig.com/t/0000001BREBFR>

2 1

fix_nated_contact and IPv6
by Sebastian Damm 27 Oct '17

27 Oct '17

Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: <sip:bob@192.168.8.132> outbound Contact: <sip:bob@1234:1234:0:1234:0:0:0:2> The correct way of transforming the Contact URI would be: Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]> This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian

4 4

Too Long Contact URI
by Soltanici Ilie 27 Oct '17

27 Oct '17

Hi All, I cannot Register some softphones because of too long Contact header when using TCP protocol. Using UDP - everything works ok and this is how the Contact header looking: sip:XXXXXXXXX@XXX.XXX.XXX.XXX:64773;rinstance=6502234ef65ea6e7 But, switching to the TCP i'm receiving the following warning: WARNING: registrar [sip_msg.c:194]: check_contacts(): contact uri is too long:[sip:XXXXXXXXX@192.168.0.1:60471;rinstance=bf4d7aae0f099bf0;transport=tcp] How can i solve this issue? Thank You.

1 1

How to send OPTIONS message by kamailio server
by 吴妙灵 27 Oct '17

27 Oct '17

Hello, I don't know how to send OPTIONS message from kamailio server to user client,can you give me same advice? Cheers, Ada

4 3

Timer function didnot trigger the routing function
by RAJA SREE RATHNA gomathinayagam 27 Oct '17

27 Oct '17

Hi, I tried using the timer function to call a function after specific time but it was not working as specified. Could you please help me with some example using timer to trigger a function. I tried it using the description given in below link. www.kamailio.org/docs/modules/devel/modules/timer.html Thanks and regards, Rathna.

2 3

Kamailio v5.0.4 Released
by Daniel-Constantin Mierla 27 Oct '17

27 Oct '17

Hello, Kamailio SIP Server v5.0.4 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.0.x versions are strongly recommended to be upgraded to v5.0.4. For more details about version 5.0.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/10/kamailio-v5-0-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 4

Set timeout before 180/183 reply arrived
by Yu Boot 27 Oct '17

27 Oct '17

Hello. Have a problem with some uplinks. They sometimes respond "100 Trying" on initial INVITE and... that's all. A caller waits for about 30 seconds and cancel a call deciding there's something wrong with our network. How to use failure route when 180/183 answers don't arrive in 5 or 10 seconds after INVITE?

2 2

Get value of a custom header param
by Arsen 26 Oct '17

26 Oct '17

Hi guys! Is it possible to get value of a custom parameter of the authorization header? I've checked selects in the textops mod and try the following: if (is_present_hf("Authorization")) { xplog("L_ERR", "custom auth $sel((a)hf_value.authorization.edatech_opaque) \n"); } Arsen Semionov www.eurolan.info cell: +442035198881

3 5

Upgrading from 1.5 to 3.x
by Brandon Shiers 26 Oct '17

26 Oct '17

I have a client that has an old Kamailio 1.5 installation. They would like to migrate and get off their old hardware and old OS. I've read the upgrade page for the 1.5.x migration to 3.0.0, are there any gotchas to be aware of other than what's there? Brandon Shiers, Network Engineer 937 West Main Street Riverton, WY 82501 307.857.6704 (o) 307.840.2366 (c) 307.856.1499 (f) brandon.shiers(a)cerento.com<mailto:Brandon.Shiers@cerento.com> [http://www.cerento.com/emailsignatures/logo.jpg]

2 1

Regarding rtimer usage under specific condition
by RAJA SREE RATHNA gomathinayagam 26 Oct '17

26 Oct '17

Hello, While going through the rtimer module we noticed that it execute the routing function repeatedly after the given interval of time. But our specification is to execute the routing function only once and timer should be enabled based on specific condition. Also there is no enable or disable option for rtimer module as in timer module, whether rtimer module can be enabled based on specific condition.If it is possible could you please suggest the solution. Thanks & Regards, Rathna.

2 1

shared dialogs
by : Paolo Visintin - Time-Net S.r.l. 26 Oct '17

26 Oct '17

Hi kamailio users, I'm wondering if there's something already done with shared dialog among more kamailio instances, in order to manage branches initalized by another kamailio instance. A ctually seems not possible (also with db mode) because kamailio is looking into caller/callee_sock and if it's different does not manage in a "standard" failover environment (master / slave) with vIP and keepalived and $fs = VIRTUAL_IP everything is working fine but in a distributed environment this could not happen as the relay is managed by a kamailio proxy Thanks in advance Paolo

4 7

I-CSCF Prefixes IMPU with + Symbol
by Ron McLeod 25 Oct '17

25 Oct '17

I am working with v5.0.3 and the IMS extensions. When I attempt to send a SIP MESSAGE from the application server towards the UE, the I-CSCF used to determine the S-CSCF for the subscriber by sending a Location-Info request to the HSS. I found that this request was failing because the IMPU (IMS Public User Identity) was being mangled by the I-CSCF, prefixing the user part with a + symbol. The example configuration file for the I-CSCF does in-fact does cause the r-uri to be prefixed with a + symbol if one is not already present. I'm wondering why since it breaks the LIR/LIA lookup. AS I-CSCF HSS | MESSAGE | | | ---------------------> | | | sip:ron@provider1.test | | | | | | | | | | LIR | | | ----------------------> | | | sip:+ron@provider1.test | | | | | | | | | DIAMETER_UNKNOWN_USER | | | <---------------------- | | | | | | | | 403 | | | <--------------------- | | | FORBIDDEN | | route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } ... Thanks, Ron

1 0

Testing for user=phone with siputils uri_param() doesn't work
by George Diamantopoulos 25 Oct '17

25 Oct '17

Hello, In my script I've had to test for the user=phone parameter in URIs. I thought uri_param() from siputils would be handy for this operation on the RURI, so I did: if ( !uri_param("user","phone") ) { add_uri_param("user=phone"); } Unfortunately this wouldn't work (uri_param returns -1), and I end up with double user=phone params in the RURI. I was able to work around this by changing the test to: if ( !($(ru{uri.uparam}) == "phone") ) { add_uri_param("user=phone"); } But I wanted to ask if this is intentional or if I should file a bug or something. Thanks. George

3 3

Planning Kamailio v5.0.4
by Daniel-Constantin Mierla 25 Oct '17

25 Oct '17

Hello, I am considering to release Kamailio v5.0.4 later this week, likely on Wednesday or Tuesday. If you are aware of any issue in branch 5.0 which is not yet reported on github.com/kamailio/kamailio issues tracker, then do it as soon as possible to have a chance to look at it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 5

usrloc module does not work with db_cluster
by Uri Shacked 25 Oct '17

25 Oct '17

Hi, I started using db_cluster module. DB are master-slave replication both ways. #!define DBURL "cluster://kamailio" when i stop the primary mysql server, all subscribers fails to register again. seems like the usrloc module does not work with the cluster. modparam("usrloc", "db_mode", 1) modparam("db_cluster", "cluster", "kamailio=>primary=9s9s;secondary=8s8s") it works OK for modules like dispatcher and htable to reload from DB. it shows the sql driver errors, but reloads from the secondary server. I need auto expire, so, p_usrloc is not a good option. working with version 4.2 any ideas ? Uri

2 1

FW: Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 25 Oct '17

25 Oct '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

1 0

Advice on taking the best path to build a Presence & IM server
by José Seabra 24 Oct '17

24 Oct '17

Hello there, I'm writing here in order to get your experiences about a Presence & IM server. I would like to know what is the best solution to follow: - SIP Simple or XMPP? I've more knowledge about SIP Simple with kamailio but i did some researches and i found several open sources unified communications software that's implement Presence and IM using an XMPP server. So what is your opinion about which is the best solution to choose, taking in considerations the following aspects: 1. scalability 2. interopeability 3. permornace 4. features(like: Personal assistance, voicemail integration, chat group, integration with social media) Should i choose Kamailio SIP Simple with xcap or an XMPP server(Prosody or ejabberd, etc) I will appreciate a lot your opinion. Thank you Regards José -- Cumprimentos José Seabra

3 4

lcr
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

(no subject)
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

How to check 4XX/5XX response in general.
by vivek＠advaitamtech.com 24 Oct '17

24 Oct '17

Hi, Is there any way to catch all 4XX and 5XX SIP error codes rather then specifying individually ie. 404, 500, 400 etc in kamailio. Thanks, Vivek.

2 2

NatHelper ignoring force_socket module parameter
by Asgaroth 24 Oct '17

24 Oct '17

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 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, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

2 5

No music on hold with mohqueue
by Fedor 24 Oct '17

24 Oct '17

I got in my kamailio.cfg: loadmodule "mohqueue.so" # --- music on hold moh ---- modparam ("mohqueue", "db_url", "mysql://kamailio:PASSWORDHERE@localhost/ kamailio") modparam ("mohqueue", "mohdir", "/etc/kamailio/moh") modparam ("mohqueue", "moh_maxcalls", 3000) Databases for mohqueue exist rttproxy configured fed@vps1:/etc/kamailio# ll /etc/kamailio/moh/ total 185424 drwxr-xr-x 2 kamailio kamailio 4096 Oct 8 09:32 ./ drwxr-xr-x 3 root root 4096 Oct 8 10:04 ../ -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.0 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.1 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.10 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.11 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.12 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.13 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.14 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.15 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.16 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.17 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.18 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.19 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.2 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.20 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.3 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.4 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.5 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.6 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.7 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.8 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.9 -rw-r--r-- 1 kamailio kamailio 8627660 Jan 2 2017 hold.wav rtpproxy configured, but no music on hold(srtp works fine) Can anybody help me, please?

2 1

Multi crashes in the same day
by igor.potjevlesch＠gmail.com 23 Oct '17

23 Oct '17

Hello ! I have multi crashes on a Kamailio 4.2.8 instance. Already 6 coredump today, same time with the same scenario. Below the output of the last coredump: (gdb) where #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 #2 0x000000000061aad2 in qm_debug_frag (qm=0x7f9a8a7aa010, f=0x7f9a8aca5968) at mem/q_malloc.c:161 #3 0x000000000061bf1c in qm_malloc (qm=0x7f9a8a7aa010, size=24, file=0x7131d0 "<core>: re.c", func=0x71483e "subst_run", line=440) at mem/q_malloc.c:388 #4 0x0000000000505e2c in subst_run (se=0x7f9a8ac08b78, input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, count=0x0) at re.c:440 #5 0x0000000000506fff in subst_str (input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, se=0x7f9a8ac08b78, count=0x0) at re.c:515 #6 0x00007f9a87ff629d in subst_uri_f (msg=0x7f9a8abfef38, subst=0x7f9a8ac08b78 "X\215\300\212\232\177", ignored=0x0) at textops.c:744 #7 0x000000000041d51b in do_action (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1094 #8 0x0000000000429ba3 in run_actions (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1583 #9 0x000000000042a208 in run_actions_safe (h=0x7fffed081380, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1648 #10 0x0000000000542994 in rval_get_int (h=0x7fffed081380, msg=0x7f9a8abfef38, i=0x7fffed080890, rv=0x7f9a8ab64ef8, cache=0x0) at rvalue.c:924 #11 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed081380, msg=0x7f9a8abfef38, res=0x7fffed080890, rve=0x7f9a8ab64ef0) at rvalue.c:1918 #12 0x000000000041cf77 in do_action (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1064 #13 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1583 #14 0x000000000041d3f6 in do_action (h=0x7fffed081380, a=0x7f9a8ab6b070, msg=0x7f9a8abfef38) at action.c:1079 #15 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab607b8, msg=0x7f9a8abfef38) at action.c:1583 #16 0x000000000042a2d0 in run_top_route (a=0x7f9a8ab607b8, msg=0x7f9a8abfef38, c=0x7fffed081380) at action.c:1669 #17 0x00007f9a891f6d0f in prepare_new_uac (t=0x7f9a6de86518, i_req=0x7f9a8abfef38, branch=0, uri=0x7fffed0814d0, path=0x7fffed0814b0, next_hop=0x7f9a8abff1a8, fsocket=0x0, snd_flags=..., fproto=0, flags=2, instance=0x7fffed0814a0, ruid=0x7fffed081490, location_ua=0x7fffed081480) at t_fwd.c:410 #18 0x00007f9a891faded in add_uac (t=0x7f9a6de86518, request=0x7f9a8abfef38, uri=0x7f9a8abff1a8, next_hop=0x7f9a8abff1a8, path=0x7f9a8abff568, proxy=0x0, fsocket=0x0, snd_flags=..., proto=0, flags=2, instance=0x7f9a8abff578, ruid=0x7f9a8abff590, location_ua=0x7f9a8abff5a0) at t_fwd.c:855 #19 0x00007f9a89201ddd in t_forward_nonack (t=0x7f9a6de86518, p_msg=0x7f9a8abfef38, proxy=0x0, proto=0) at t_fwd.c:1723 #20 0x00007f9a891ee374 in t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, proto=0, replicate=0) at t_funcs.c:355 #21 0x00007f9a8922bd89 in _w_t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, force_proto=0) at tm.c:1517 #22 0x00007f9a8922ceee in w_t_relay (p_msg=0x7f9a8abfef38, _foo=0x0, _bar=0x0) at tm.c:1718 #23 0x000000000041d48d in do_action (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1088 #24 0x0000000000429ba3 in run_actions (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1583 #25 0x000000000042a208 in run_actions_safe (h=0x7fffed0848f0, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1648 #26 0x0000000000542994 in rval_get_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, i=0x7fffed082668, rv=0x7f9a8a83eb60, cache=0x0) at rvalue.c:924 #27 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082668, rve=0x7f9a8a83eb58) at rvalue.c:1918 #28 0x0000000000546fc2 in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082af0, rve=0x7f9a8a83f280) at rvalue.c:1926 #29 0x000000000041cf77 in do_action (h=0x7fffed0848f0, a=0x7f9a8a83fbd0, msg=0x7f9a8abfef38) at action.c:1064 #30 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a838c88, msg=0x7f9a8abfef38) at action.c:1583 #31 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996598, msg=0x7f9a8abfef38) at action.c:712 #32 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a9955b8, msg=0x7f9a8abfef38) at action.c:1583 #33 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996a50, msg=0x7f9a8abfef38) at action.c:1079 #34 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a8fa068, msg=0x7f9a8abfef38) at action.c:1583 #35 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:712 #36 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:1583 #37 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830db8, msg=0x7f9a8abfef38) at action.c:1079 #38 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38) at action.c:1583 #39 0x000000000042a2d0 in run_top_route (a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38, c=0x0) at action.c:1669 #40 0x00000000005095b6 in receive_msg ( buf=0xa78b60 "INVITE sip:0123456789@A.B.C.181:5060;transport=UDP;user=phone SIP/2.0\r\nFrom: <sip:0987654321@A.B.C.148:5060>;tag=BN1507883839-1-1507047459-332794206;tran sport=UDP\r\nTo: <sip:0123456789@H.G.F.D"..., len=1119, rcv_info=0x7fffed084be0) at receive.c:216 #41 0x000000000060945a in udp_rcv_loop () at udp_server.c:521 #42 0x00000000004a6e62 in main_loop () at main.c:1629 #43 0x00000000004acfa6 in main (argc=7, argv=0x7fffed085018) at main.c:2581 Regards, Igor.

2 2

Presence WARNING after publish and no NOTIFY
by Yuriy Gorlichenko 23 Oct '17

23 Oct '17

Hi use 5.0.1 with presence for TCP/UDP endpoints presence works PERFECT but for WS socket presence i see next (small cleaned log here with all described bellow https://pastebin.com/8a77ggj0 ): When WS endpoint subscribes - kamailio After kamailio sends 202 to SUBSCRIBE message (see it on client) it sends fast NOTIFY correcly without any trouble, and i see at the kamailio reply from WS endpoint - 200 WS endpoint added to active_watchers table. But when Subscribed endpoint makes for example some call and presence server sends PUBLISH i see: presence [presentity.c:426]: delete_presentity_if_dialog_id_exists(): Presentity already exists - deleting it At the logs and then presence server doesn't send NOTIFY to WS endpoint What is the issue of this? Can not understand why it happens. Can someone to help me understand this? And the dump i see that kamailio send first NOTIFY form UDP port but then translates it to tls for websocket and i see correct NOTIFY at the WS endpoint

2 4

MoS(QoS) measurement in Kamailio based IMS network
by Kranti Kumar 23 Oct '17

23 Oct '17

Hi All, I am using Kamailio IMS VMimage which is provided by Franz. I am successfully established the call between the IMS clients. Here I would like to measure the MoS mean opnion score (QoS) of Calls (Voice quality). Can any one suggest me / guide me how to measure the same in this IMS network. Thank you so much in advance. Regards, -kranti

2 1

how to bridge two landline users
by 赵国杰 23 Oct '17

23 Oct '17

Hello, In a standard sip flow, the call goes like: sip user A --> kamailio --> pstn --> landline user B. However, when user A has a bad internet access, the audio is broken. So what I want is to let sip user A send a invite to kamailio first, then kamailio send invite to user A and B's landline number through pstn, then bridge the two call together. I understand this can be achieved by using FREESWITCH originate and bridge command. I've tried but there's no audio both ways, which really makes me feel stupid of myself. So I'm wondering if this can be done with kamailio? If so, how? Thanks Jesse

2 1

uac_replace_from and header adulterations
by Alex Balashov 23 Oct '17

23 Oct '17

Hello, I've got a bizarre problem caused by bad UA behaviour: UA A ---> Kamailio (P) ---> UA B 1. UA A sends initial INVITE through P to B; 2. Kamailio (P) makes some modifications to the From header using uac_replace_from() and passes along to B. 3. B sends an in-dialog request (e.g. BYE or reinvite) to A through P; in doing so, it modifies the To (formerly From) value slightly, replacing the hostname portion in the To URI with a different value to the one that was received in the From header. 4. Kamailio relays this in-dialog request to A, but with an adulterated/clipped/truncated/grammatically invalid To header now. 5. A responds with 400 Bad Request due to invalid To header. I would not dispute that UA B should not be modifying the remote URI in this manner. But since it does, it gives rise to two questions: 1. Why does Kamailio not simply discard the modified To header and restore the original value stored in the Record-Route rider parameter? Is it because the Record-Route parameter does not contain the original header value, but rather some data complementary to the current header value? Or is it because the UAC code takes a checksum of the original remote URI header value and stores it, and just checks it when restoring on principle? If so, what's the motive for that? 2. Wouldn't it be better behaviour to simply reject a request so malformed, rather than passing it on with a corrupt restored value? If Kamailio can detect that the header has been tampered with, why pass it on? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Help on Video Conference
by Kranti Kumar 23 Oct '17

23 Oct '17

Hello , I’m trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ My configuration details: HSS - 10.0.0.9 (./hss.sh) PCSCF - 10.0.0.10 (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg) ICSCF - 10.0.0.11(kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg) SCSCF - 10.0.0.12 (kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg) host machine: 10.0.0.5 I am able to register mobile clients ( IMSdroid) and make voice/video calls between them successfully. Now I would like to make a conference call among 3 IMSdroid users but its failing. Could you please suggest me the way forward here. I am thinking to integrate Application server (AS) to enable Video Conferences. Could you please guide me how to enable AS in this VM image and configuration details. Is there any possibility to do Video conference using this current setup(VM image). Could you please suggest me. Thank you so much in advance. Regards, -kranti

2 1

Kamailio IMS deployment
by Ramya Y 22 Oct '17

22 Oct '17

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

5 22

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