- sr-users - kamailio.org

FHoSS db web interface failing for retrievals
by Woscek, Martin W. 18 Jun '19

18 Jun '19

Hi all, We can enter user data into our HSS, we just cant retrieve data. Just to verify, FHoSS is based on Java 1.7 and java.lang.CharSequence is in Java 1.7? What we have: java version "1.7.0_80" Java(TM) SE Runtime Environment (build 1.7.0_80-b15) Java HotSpot(TM) 64-Bit Server VM (build 24.80-b11, mixed mode) The error: HTTP Status 500 - ________________________________ type Exception report message description The server encountered an internal error () that prevented it from fulfilling this request. exception org.apache.jasper.JasperException: Unable to compile class for JSP An error occurred at line: 116 in the jsp file: /pages/user/imsu_search_result.jsp Generated servlet error: The type java.lang.CharSequence cannot be resolved. It is indirectly referenced from required .class files org.apache.jasper.compiler.DefaultErrorHandler.javacError(DefaultErrorHandler.java:84) org.apache.jasper.compiler.ErrorDispatcher.javacError(ErrorDispatcher.java:328) org.apache.jasper.compiler.JDTCompiler.generateClass(JDTCompiler.java:397) org.apache.jasper.compiler.Compiler.compile(Compiler.java:288) org.apache.jasper.compiler.Compiler.compile(Compiler.java:267) org.apache.jasper.compiler.Compiler.compile(Compiler.java:255) org.apache.jasper.JspCompilationContext.compile(JspCompilationContext.java:556) org.apache.jasper.servlet.JspServletWrapper.service(JspServletWrapper.java:293) org.apache.jasper.servlet.JspServlet.serviceJspFile(JspServlet.java:291) org.apache.jasper.servlet.JspServlet.service(JspServlet.java:241) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) org.apache.struts.action.RequestProcessor.doForward(RequestProcessor.java:1063) org.apache.struts.tiles.TilesRequestProcessor.doForward(TilesRequestProcessor.java:263) org.apache.struts.action.RequestProcessor.processForwardConfig(RequestProcessor.java:386) org.apache.struts.tiles.TilesRequestProcessor.processForwardConfig(TilesRequestProcessor.java:318) org.apache.struts.action.RequestProcessor.process(RequestProcessor.java:229) org.apache.struts.action.ActionServlet.process(ActionServlet.java:1194) org.apache.struts.action.ActionServlet.doPost(ActionServlet.java:432) javax.servlet.http.HttpServlet.service(HttpServlet.java:709) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) Thanks, _Martin

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Kamailio Crashing when using Evapi with CGRateS
by Efelin Novak 18 Jun '19

18 Jun '19

Hi folks, I am doing some experiments with Kamailio and Cgrates. To be precise I am testing evapi module. When in testing environment I do not have any problem. After redirecting some production traffic to this kamailio, It crashes in at random times (few minutes to 20). I didn't find any pattern here. CGRateS doesn't write anything in logs. I'm on the last 4.4 Kamailio (compiled recently). CGRateS is on the last version. Can you please point to how to continue with debugging? Thanks Efelin Here is the problem in syslog: kamailio[9594]: : <core> [mem/f_malloc.c:599]: fm_free(): BUG: fm_free: bad pointer 0x97894818 (out of memory block!), called from tm: h_table.c: free_cell_helper(185) - abortingkamailio[9608]: CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 16 kamailio[9588]: ALERT: <core> [main.c:743]: handle_sigs(): child process 9594 exited by a signal 6 kamailio[9588]: ALERT: <core> [main.c:746]: handle_sigs(): core was not generated kamailio[9588]: INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD kamailio[9604]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received kamailio[9606]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received kamailio[9603]: INFO: <core> [main.c:823]: sig_usr(): signal 15 received bt full #0 0xb77d1424 in __kernel_vsyscall () No symbol table info available. #1 0xb7673661 in raise () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7676a92 in abort () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 No symbol table info available. #3 0x082611a3 in fm_free (qmp=0x56779000, p=0x2d4e4f49, file=0x970932ee "tm: h_table.c", func=0x970935b7 "free_cell_helper", line=185, mname=0x97092d20 "tm") at mem/f_malloc.c:601 qm = 0x56779000 f = 0x56df046c __FUNCTION__ = "fm_free" #4 0x96fd0559 in free_cell_helper (dead_cell=0x56d935c4, silent=0, fname=0x970b90ca "timer.c", fline=648) at h_table.c:185 b = 0x56df0490 "CANCEL sip:00444123456@192.168.1.3:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKb6ae.5d03d7e94e5ec25f34930629f434a006.1\r\nMax-Forwards: 66\r\nFrom: \"0044454321\" <sip:0044454321@i"... i = 1 rpl = 0x9797e818 tt = 0xbfe44c68 foo = 0x970ccb14 cbs = 0x0 cbs_tmp = 0x56d12b68 __FUNCTION__ = "free_cell_helper" #5 0x9706884f in wait_handler (ti=31715179, wait_tl=0x56d9360c, data=0x56d935c4) at timer.c:648 p_cell = 0x56d935c4 ret = 0 #6 0x08241e20 in timer_list_expire (t=31715179, h=0x567a3b68, slow_l=0x567a5394, slow_mark=1772) at timer.c:873 tl = 0x56d9360c ret = 0 #7 0x08242211 in timer_handler () at timer.c:938 saved_ticks = 31715179 run_slow_timer = 0 i = 748 __FUNCTION__ = "timer_handler" #8 0x0824266b in timer_main () at timer.c:977 No locals. #9 0x080f0005 in main_loop () at main.c:1699 i = 3 pid = 0 si = 0x0 si_desc = "udp receiver child=2 sock=192.168.1.2:5060\000\270\301\310V\270\301\310V\000\000\000\000\000\000\000\020c\000\000\000\001\000\000\000\000\000\000\000\374Q\344\277HO\344\277\341`&\b\230\071\220\227\004\000\000\000٧5\b$\271\065\bn\000\000\000\324\247\065\b\n\000\000\000\363\320\372\060", '\000' <repeats 11 times> nrprocs = 3 woneinit = 1 __FUNCTION__ = "main_loop" #10 0x080f6b3a in main (argc=13, argv=0xbfe45374) at main.c:2644 cfg_stream = 0x8491008 c = -1 r = 0 tmp = 0xbfe4577c "" tmp_len = 1 port = 0 proto = 0 options = 0x8350e74 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:" ret = -1 seed = 1759582648 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbfe452a0 p = 0x805cc8c "[\201Ä4;" st = {st_dev = 14, __pad1 = 0, st_ino = 6161, st_mode = 16877, st_nlink = 2, st_uid = 109, st_gid = 111, st_rdev = 0, __pad2 = 0, st_size = 60, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1560160293, tv_nsec = 983219454}, st_mtim = {tv_sec = 1560165082, tv_nsec = 861178334}, st_ctim = {tv_sec = 1560165082, tv_nsec = 861178334}, __unused4 = 0, __unused5 = 0} __FUNCTION__ = "main"

2 2

Modify DMQ 200OK in on_reply route
by jenus＠cyberchaos.nl 18 Jun '19

18 Jun '19

Hi all, I'm trying to run a kamailio DMQ node inside a docker container. All works fine but i have one issue with DMQ. Since the kamailio process inside the container is behind a nat i have set the advertise address to the external ip, this works fine for the via headers but DMQ keeps sending the internal interface in the KDMQ notification_peer messages. The external ip is in there as well. Example: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.13;branch=z9hG4bK2dcb.c6326de2000000000000000000000000.0 To: <sip:notification_peer@10.10.10.94:5060>;tag=616b7acb681ed5e9cbc485b3faa09d62.0a3d From: <sip:notification_peer@10.10.10.13:5060>;tag=e8f29c06c4b85a188f6533f08d60feb6-ff29 CSeq: 10 KDMQ Call-ID: 17042656602ca532-23367(a)x.x.x.x Content-Type: text/plain Server: kamailio (5.1.4 (x86_64/linux)) Content-Length: 222 sip:10.10.10.13:5060;status=active sip:10.10.10.12:5060;status=active sip:10.10.10.92:5060;status=active sip:10.10.10.93:5060;status=active sip:10.10.10.94:5060;status=active <-- entry for container external ip sip:172.17.0.2:5060;status=disabled <-- entry for container internal ip I tried to strip this entry using the replace_body_atonce in the tm:local-request route. This works fine for the KDMQ request, but when i try to remove the same line from 200OK responses that are send out as a response to incomming KDMQ messages this seems to fail. I use the following code: # Working code (for outgoing KDMQ requests): event_route [tm:local-request] { if(is_method("KDMQ") && $rU == "notification_peer") replace_body_atonce("sip:172.17.0.*:5060;status=.*", ""); } # Failing code (for 200OK responses): reply_route { xlog("L_ALERT", "REPLY: Sending out reply, tU is $tU rs is $rs\n"); if ($rs == "200" && $tU == "notification_peer") { replace_body_atonce("sip:172.17.0.*:5060;status=.*", ""); } } I'm running kamailio 5.1.4. Is there a way to strip a line from the body of a local generated 200OK response? Or is there a way to avoid that DMQ inserts the internal container ip in the first place. Thanks, Jan Hazenberg

3 11

Rabbitmq module blocks sip request
by Denys Pozniak 18 Jun '19

18 Jun '19

Hello! I use rabbitmq module to get destination IP for the next call routing, but this module blocks sip request in case of RabbitMQ server is unreachable (timeout_sec/usec does not run out here). - Is there any solution to avoid such behavior and force function rabbitmq_publish_consume() to return an error code and continue script execution? - Is this connected with the nature of rabbitmq-c client? ("Currently librabbitmq offers no async API, but sync API, with a timeout.") -- BR, Denys Pozniak

1 0

Info: Kamailio.org - Infrastructure Maintenance - June 18, 2019
by Daniel-Constantin Mierla 18 Jun '19

18 Jun '19

The infrastructure hosting the main server for Kamailio project will go under scheduled maintenance during June 18, 2019 (tomorrow). The work is expected to start around 7:00am UTC and can last for a few hours. During the maintenance interval, the services related to Kamailio project may become unavailable, among them: * the web server (www.kamailio.org), including the wiki portal (www.kamailio.org/wiki) * the mailing lists and the email contact addresses * the GIT repository mirror git.kamailio.org Once the maintenance work is completed, there will be an update notification sent about it! If there are relevant events to be notified, we are going to publish on our twitter stream: * https://twitter.com/kamailio Also, you can join #kamailio channel on irc.freenode.net in case you want to report some incident related to our infrastructure that you discover while maintenance goes on. For bugs or other issues related to Kamailio code, use the tracker on github.com/kamailio/kamailio – that service is not affected being maintained by GitHub corporation. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

Kamailio 4.4.x CDR db accounting
by Laura 18 Jun '19

18 Jun '19

Dear guys, I'm looking to understand the "info" field inside the CDR written on acc table for Kamailio 4.4.x.. It seem that the field is never written with datas.. Is there a way to write some informations inside it ? Thanks for any reply

3 3

Forward calls between kamailio registrar nodes (dmq + dmq_usrloc)
by Ildar Mukhametzhanov 18 Jun '19

18 Jun '19

Hello! I am developing setup with several kamailio nodes as registrar cluster (dmq + dmq_usrloc). Location sync works, I see locations of both servers on every server. Is there any standard approach to redirect call between nodes except replacing of $ru? Executing of lookup("location") for user registered on another node return negative response. -- Sincerely, *Ildar Mukhametzhanov* Head of software development VoiceON Telecom Services GmbH im(a)voice-on.net <mailto:im@voice-on.net> t. +49 30 398 20 500-77

1 0

Authenticating xhttp request
by Olli Attila 18 Jun '19

18 Jun '19

Hello, I have this xhttp event_route on Kamailio that I am using to signal the proxy to reload dialplans and htable when necessary: event_route[xhttp:request] { if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Now instead of returning 403 forbidden for requests coming from other src_ip than proxy itsef, I would like to authenticate the http request via proxy database. How can this be done if possible? Cheers, Olli

3 7

dispatcher and TLS targets
by Karsten Horsmann 17 Jun '19

17 Jun '19

Hi all, i try to configure an SBC OS config [1] based kamailio 5.2.3 [2] with dispatcher and rtpengine. I used transport=tcp to see the plain traffic and then switched to TLS (with tls.cfg, valid certificate and stuff). After starting up, the Target is marked as "down". Due the encryption its hard to debug that. Any hints? Did i made an mistake in the configuration? TLS calls from the target to my kamailio proxy works. So its "half broken" :) at the moment. [1] https://github.com/voiceboys/sbcOS/blob/master/SbcOS/configs/voice/kamailio… [2] kamailio -v version: kamailio 5.2.3 (x86_64/linux) c36229 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, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c36229 compiled on 11:28:11 May 22 2019 with gcc 4.8.5 -- %< --------------------- kamctl dispatcher dump "SET": { "ID": 1004, "TARGETS": [{ "DEST": { "URI": "sip:sip101.example.de;transport=tls", "FLAGS": "TP", "PRIORITY": 0, "ATTRS": { "BODY": "access=212.xx.xx.xx:5061;socket=tls:212.xx.xx.xx:5061;weight=100;ping_from=sip: mykamailio.example.de", "DUID": "", "MAXLOAD": 0, "WEIGHT": 100, "RWEIGHT": 0, "SOCKET": "tls:212.xx.xx.xx:5061" }, "LATENCY": { "AVG": 30000, "STD": 0, "EST": 30000, "MAX": 30000, "TIMEOUT": 1 } } }] } }, -- %< --------------------- kamctl dispatcher dump WARNING: <script>: Destination down: OPTIONS ru=sip101.example.de;transport=tls du=<null> -- %< --------------------- tls.cfg [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de [server:212.xx.xx.xx:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client: 212.xx.xx.xx:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/mykamailio.example.de.pem certificate = /etc/pki/tls/private/mykamailio.example.de.pem server_name = mykamailio.example.de [client:default] verify_certificate = no require_certificate = no -- %< --------------------- tls.cfg Cheers Karsten -- Mit freundlichen Grüßen *Karsten Horsmann*

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sr -user unsubscribe request
by Martin Lane 17 Jun '19

17 Jun '19

Please unsubscribe me from the sr-users list. V/R, Martin

1 0

Dispatcher's probing malfunction
by Duarte Rocha 17 Jun '19

17 Jun '19

Greetings, I have some questions about dispatcher's probing mechanics : 1 - In my SBC i have 26 gateways being probed. However, if i do a SIP capture no more than 16 gateways are probed each minute. Some gateways are always probed, others only once in a while. 2 - I have two SBC working in a pair with DB being replicated and with HA IPs. That being said i configure one NAP twice, each configuration with one HA socket. Is this configuration supported? I get very weird behaviours when i configure it like this. Best Regards, Duarte Rocha

1 0

restrict calls to only registered users
by Gaurav Bmotra 17 Jun '19

17 Jun '19

hi i m using kamailio 5.1 , with UBUNTU 18.1 i want to restrict call to only registered user ,,, plz help me thank you -- *Regards:* Gaurav Kumar

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Querying in-memory location table (dmq_usrloc)
by Gholamreza Sabery 16 Jun '19

16 Jun '19

I am using dmq_usrloc, to replicate user locations across multiple Kamailio instances. Everything works fine. The problem occurs when a user on Kam1 wants to call a user on Kam2. How can I query the in-memory location table to see where the user is registered? PS: I use a Path header to specify the server that each user is registered on.

2 2

Softswitch
by Mary Honey 16 Jun '19

16 Jun '19

Hello, Is it possible to use Kamailio as a grade 5 Softswitch ? I need to handle 1000 calls/s

2 1

Stateless message forwarding with forward() add extra identical Via header.
by Ali Shirvani 15 Jun '19

15 Jun '19

Hi all, According to this SO question <https://stackoverflow.com/questions/32519551/kamailio-concurrent-calls-afte…> I tried to forward received message to multiple destinations with the following Lua script. But in the second UA I got duplicate Via header. Here is the Lua script: ```Lua KSR.forward_uri("sip:ali@192.168.1.50:7080;transport=UDP"); KSR.forward_uri("sip:tes@192.168.1.50:9080;transport=UDP"); ``` And here is the received Via headers is `tes` UA: ``` Via: SIP/2.0/UDP 192.168.1.21;branch=z9hG4bKbe8c.953ed42f0bcc438c6991121d3483010b.0 Via: SIP/2.0/UDP 192.168.1.21;branch=z9hG4bKbe8c.953ed42f0bcc438c6991121d3483010b.0 Via: SIP/2.0/UDP 10.189.219.4:5060 ;rport=5060;rport=5060;branch=z9hG4bK-4177-1-2 ``` Would you please help me to forward message to multiple destination without duplicate Via headers? Regards, Ali

1 0

403 Forbidden
by Pejic, Miroslav 14 Jun '19

14 Jun '19

Hi :) I'm trying to save a failed call in the kamailio_router.acc_cdrs database after sending the 403 Forbidden by router, using the module acc, if is it possible? https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html Something like this: route[IS_CALLER_BLACKLISTED] { python_exec("check_caller_blacklisted"); if ($var(callerBlacklisted) == 1){ t_send_reply("403", "Forbidden"); IT’S POSSIBLE HERE SAVE A CALL ON KAMAILIO_ROUTER.ACC_CDRS? exit; } Thanks in advance! Miro

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Re: [SR-Users] 403 Forbidden
by Federico Cabiddu 14 Jun '19

14 Jun '19

I understand, to get the cdr you should have created the dialog before replying. Cheers, Federico On Fri, 14 Jun 2019, 17:05 Pejic, Miroslav, <miroslav.pejic(a)cloud.timenet.it> wrote: > [image: image.png] > > The call doesn't even begin, I immediately block it. :) > > Il giorno ven 14 giu 2019 alle ore 16:59 Pejic, Miroslav < > miroslav.pejic(a)cloud.timenet.it> ha scritto: > >> Hi Federico, >> >> i tryied also with `modparam("acc", "cdr_on_failed", 1)` >> but in my case the call is not saved at all on `acc_cdrs`, >> with the second method i mentioned works :) >> Thanks >> Miro >> >> Il giorno ven 14 giu 2019 alle ore 16:14 Federico Cabiddu < >> federico.cabiddu(a)gmail.com> ha scritto: >> >>> Hi, >>> dialog CDR for failed calls can be generated enabling >>> >>> http://www.kamailio.org/docs/modules/devel/modules/acc.html#acc.p.cdr_on_fa… >>> >>> It has been some time since I last tested it, but it still should work :) >>> >>> Cheers, >>> >>> Federico >>> >>> On Fri, Jun 14, 2019 at 3:25 PM Daniel-Constantin Mierla < >>> miconda(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> it should be ok also with 2) if it gives what you want. Kamailio is >>>> just writing to acc_cdrs, if you have extra columns, it should not be >>>> affected. >>>> >>>> Cheers, >>>> Daniel >>>> On 14.06.19 14:16, Pejic, Miroslav wrote: >>>> >>>> Hi Daniel :) >>>> >>>> i tryied 2 methods to save it >>>> >>>> 1) i created a new table on kamailio_router called `acc` >>>> >>>> from `utils/kamctl/mysql/acc-create.sql` >>>> >>>> ``` >>>> CREATE TABLE `acc` ( >>>> `id` INT(10) UNSIGNED AUTO_INCREMENT PRIMARY KEY NOT NULL, >>>> `method` VARCHAR(16) DEFAULT '' NOT NULL, >>>> `from_tag` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `to_tag` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `callid` VARCHAR(255) DEFAULT '' NOT NULL, >>>> `sip_code` VARCHAR(3) DEFAULT '' NOT NULL, >>>> `sip_reason` VARCHAR(128) DEFAULT '' NOT NULL, >>>> `time` DATETIME NOT NULL >>>> ); >>>> CREATE INDEX callid_idx ON acc (`callid`); >>>> ``` >>>> then >>>> >>>> ``` >>>> route[IS_CALLER_BLACKLISTED] { >>>> >>>> python_exec("check_caller_blacklisted"); >>>> >>>> if ($var(callerBlacklisted) == 1){ >>>> >>>> t_send_reply("403", "Forbidden"); >>>> >>>> acc_db_request("403 Forbidden", "acc"); >>>> >>>> exit; >>>> >>>> } >>>> >>>> ``` >>>> >>>> in `acc` i have now calls that were not answered with 200ok :) >>>> >>>> >>>> 2) add new fields on exsisting acc_cdrs table used to store cdrs >>>> >>>> ``` >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `from_tag` varchar(128) DEFAULT NULL >>>> AFTER `dst_domain`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `to_tag` VARCHAR(128) DEFAULT NULL >>>> AFTER `from_tag`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `sip_code` VARCHAR(3) DEFAULT NULL >>>> AFTER `to_tag`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `sip_reason` varchar(128) DEFAULT >>>> NULL AFTER `sip_code`; >>>> ALTER TABLE `acc_cdrs` ADD COLUMN `time` datetime DEFAULT NULL AFTER >>>> `sip_reason`; >>>> ``` >>>> >>>> ``` >>>> route[IS_CALLER_BLACKLISTED] { >>>> >>>> python_exec("check_caller_blacklisted"); >>>> >>>> if ($var(callerBlacklisted) == 1){ >>>> >>>> t_send_reply("403", "Forbidden"); >>>> >>>> acc_db_request("403 Forbidden", "acc_cdrs"); >>>> >>>> exit; >>>> >>>> } >>>> >>>> ``` >>>> >>>> In this way i have both stored calls that were not answered with 200ok >>>> and that were answered with 200ok. >>>> The idea is to have all in one table `acc_cdrs`. >>>> >>>> Can be ok also the second solution? >>>> >>>> Thank you very much! >>>> Miro >>>> >>>> Il giorno ven 14 giu 2019 alle ore 09:06 Daniel-Constantin Mierla < >>>> miconda(a)gmail.com> ha scritto: >>>> >>>>> Hello, >>>>> >>>>> I don't think that you can get a dialog-based CDR for calls that were >>>>> not answered with 200ok. But you can get a record in acc table, by storing >>>>> failed transaction record (see the mod params for acc and failed >>>>> transaction flag or use acc_db_request()). >>>>> >>>>> Cheers, >>>>> Daniel >>>>> On 13.06.19 22:45, Pejic, Miroslav wrote: >>>>> >>>>> Hi :) >>>>> >>>>> >>>>> I'm trying to save a failed call in the kamailio_router.acc_cdrs >>>>> database >>>>> >>>>> after sending the 403 Forbidden by router, using the module acc, if is >>>>> it possible? >>>>> >>>>> >>>>> https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html >>>>> >>>>> >>>>> Something like this: >>>>> >>>>> >>>>> route[IS_CALLER_BLACKLISTED] { >>>>> >>>>> >>>>> python_exec("check_caller_blacklisted"); >>>>> >>>>> >>>>> if ($var(callerBlacklisted) == 1){ >>>>> >>>>> >>>>> t_send_reply("403", "Forbidden"); >>>>> >>>>> >>>>> IT’S POSSIBLE HERE SAVE A CALL ON KAMAILIO_ROUTER.ACC_CDRS? >>>>> >>>>> >>>>> >>>>> exit; >>>>> >>>>> } >>>>> >>>>> >>>>> Thanks in advance! >>>>> >>>>> Miro >>>>> >>>>> _______________________________________________ >>>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>> >>>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>

1 0

Getting which timer has expired
by Igor Olhovskiy 14 Jun '19

14 Jun '19

Hi! Is there any way to get which timer on transaction actually expired? Scenario is following: INVITE - 180 - 200 OK - (restransmissions 250 msec starts) - pause 15 sec - ACK. In INVITE i'm setting t_on_reply("MY_REPLY") But in around 10 sec after first 200 OK (and 12 sec after INVITE) I'm getting in logs DEBUG: tm [h_table.c:133]: free_cell_helper(): freeing transaction 0x7fb632d77090 from timer.c:689 and on next 200 OK I'm never getting into onreply_route[MY_REPLY] again. Nowhere in the code or config I have explicit set any of timers with 10 or 12 seconds value. Most of them are default. So, can I somehow get which timer expired exactly? PS: If any, pause in 15 sec for a reason to emulate this issue. Regards, Igor

2 2

Event Route not found when loading XHTTP module
by Dhruva TS 14 Jun '19

14 Jun '19

Hello, I am new to Kamailio and I am using Kamailio 5.2. When trying to the load the XHTTP module, event route not found error is thrown and XHTTP is module is not getting loaded. I am trying to use JSONRPC module through HTTP transport for RPC calls. Using XMLRPC, things work fine. DEBUG: <core> [core/sr_module.c:896]: init_mod(): xhttp ERROR: xhttp [xhttp_mod.c:136]: mod_init(): failed to find event_route[xhttp:request] ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module xhttp (/usr/lib/kamailio/modules/xhttp.so) ERROR: error while initializing modules I have added below to Kamailio.cfg file: loadmodule "xhttp.so" event_route[xhttp:request] { if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Any help in this regard. Thanks, Dhruva

2 1

S-CSCF replies “Forbidden - Domain not served”
by Bao, Yin 13 Jun '19

13 Jun '19

We have configured Kamailio IMS (P, I, S-CSCF) to use a single realm with the same NETWORKNAME and trying to make a call between bob and alice. We are getting S-CSCF replies “Forbidden – Domain not served” in two scenarios: 1. When bob calls alice, the INVITE went from P->S->I->S and the destination S-CSCF responded this error 2. During SIP registration in responding to SUBSCRIBE (endpoint still can register even with this reply, so we proceeded with making call) In quick search we found that in scscf/kamailio.cfg route{ } implementation there is a case when this error is generated: # Terminating if (uri == myself || uri =~ "tel:.*") { if (!term_impu_registered("location")) { xlog("L_DBG", "We need to do an UNREG server SAR assignemnt"); assign_server_unreg("UNREG_SAR_REPLY", "location", "term"); exit; } } else { sl_send_reply("403","Forbidden - Domain not served"); exit(); } route(term); break; What is this error designed for? Also in general are we supposed to rework the routing code in Kamailio.cfg (P, S, I-CSCF)? Thanks for any suggestions in advance. Yin

1 0

Problem with Kamailio 5.2.x Repository
by Joshua Riffle 13 Jun '19

13 Jun '19

Hi, Heads up... I suspect the Kamailio 5.2.x RPM files for CentOS 7 were recently updated June 10th and are now no longer accepted by a standard yum install... *Repository file in /etc/yum.repos.d/:* [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl= http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… gpgcheck=1 gpgkey= http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… enabled=1 *Operating System Version:* CentOS Linux release 7.6.1810 (Core) (Fresh image -- varying cloud providers) *Command to install:* yum install kamailio *Resulting error message:* kamailio-5.2.3-4.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.3-4.el7.centos.x86_64: [Errno 256] No more mirrors to try. *Troubleshooting:* I verified the IP address for the opensuse URL with three different Internet providers and tested the installation on servers in different cloud providers geolocated in both Canada and United States and received the exact same results in all cases. -- Joshua Riffle *DevOps Lead & Duke of the Dial Tone* We Are Skyetel and You Are Awesome. p: (360) 843-9903 o: (360) 986-5200 s: http://www.skyetel.com e: josh(a)skyetel.com

2 2

CRITICAL ERROR in data_lamp.c!
by Mojtaba 13 Jun '19

13 Jun '19

Hello, What does this CRITICAL error mean? Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: <core> [core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out of memory block!) called from core: core/data_lump.c: free_lump(466) - aborting I got it when i make new call during debugging new module in Kamailio? In module we have to change some message's body with data_lump. I asked relaed issue (strange behaviour while developing new module) in Kamailio developer forum <sr-dev(a)lists.kamailio.org> With Best Regards. -- --Mojtaba Esfandiari.S

2 10

dialog:end event route
by Igor Olhovskiy 13 Jun '19

13 Jun '19

Hi! Just to confirm, is event route [dialog:end] not called on fr_inv_timer expiry? Cause it's called on call busy (486) and reject (whatever 6xx). So, what's the idea of not calling it on timeout? Or I'm missing something? Kamailio 5.2 Thanks! Regards, Igor.

2 8

Best practises for websocket authentication
by Grant Bagdasarian 12 Jun '19

12 Jun '19

Hello, I was wondering if there are any best practices for authenticating websocket connection establishment inside the xhttp event route? Thanks and regards, Grant

3 3

TCP reset behavior
by R, Rajkumar (Raj) 12 Jun '19

12 Jun '19

Hi All, In our setup running kamailio (alpine image) in stateless mode and have following TCP config set. After client registers to the backend see two TCP connections created - one towards the client and the other towards the backendserver. If there's a abrupt network disconnect either at the client side or at the backend side, do not see TCP reset happening from kamailio as per the config. Per below config expected after 120seconds( = [tcp_keepidle] + ( [tcp_keepintvl * tcp_keepcnt] ) instead it keeps resending TCP keepalive packets at varying intervals for varying duration length(like 6mins, 14mins..) before resetting the TCP connections. Also want to mention that we are not using TCPOPS module here. I am trying to achieve consistent duration for TCP reset in kamailio. Could you please point me if I have missed anything here? Setup: Client ---- Kamailio (stateless mode) ---- Backend server TCP Config: tcp_connection_lifetime = 3600 tcp_accept_aliases = no tcp_async = yes tcp_connect_timeout = 10 tcp_crlf_ping = yes tcp_keepalive = yes tcp_keepcnt = 3 tcp_keepidle = 30 tcp_keepintvl = 30 # Total 2 mins - 30+(30*3)=120 sec tcp_linger2 = 30 tcp_rd_buf_size = 65536 tcp_send_timeout = 10 thanks, raj

2 9

Setting the probing state failed
by Daniel Greenwald 12 Jun '19

12 Jun '19

ERROR: dispatcher [dispatch.c:2635]: ds_options_callback() Setting the probing state failed (sip:10.0.9.133:5064, group 0) I get this message periodically from 3 kamailio dispatchers simultaneously for the same destination host. Any idea what this error means? Dispatcher settings below: modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "sock_avp", "$avp(dssocket)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "ds_hash_size", 3) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:pinger@MYPRIVATEIP :INTERNALPORT") #modparam("dispatcher", "ds_ping_method", "INFO") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404")

2 1

How reply INVITE request with 200 OK that contains SDP?
by Ali Shirvani 11 Jun '19

11 Jun '19

Hi all, I tried to implement simple announcer by using play_media function in latest rtpengine module. I implemented simple Lua script and I tried to test the implementation with simple SIPp scenario. Here is the Lua function to reply the received INVITE request. ```Lua function ksr_route_announcer() KSR.info("======= Announcer =====\n"); local u = KSR.pv.gete("$rU"); KSR.info("username is: " .. u .. "\n"); if u == "ann" then KSR.info("announcer has been called.\n"); if KSR.textops.has_body_type("application/sdp") then KSR.info("request contains SDP.\n"); KSR.tm.t_newtran(); KSR.rtpengine.rtpengine_manage0(); KSR.tm.t_reply(200, "OK"); KSR.x.exit(); end end end ``` According to kamailio logs rtpengine modified the INVITE SDP, but I want to add SDP to 200 OK response. How should I manage rtpengine to append SDP answer to the 200 OK response? The attached is the SIPp log, as you can see that last 200 OK doesn't contain SDP. Regards, Ali

2 1

Re: [SR-Users] [EXT] Re: Kamailio 5.2.2 ICSCF changing sip to tel
by Woscek, Martin W. 11 Jun '19

11 Jun '19

Daniel, Actually, the HSS is not returning the uri with the +. There are no users int the HSS database with a public ID with a + so the HSS returns the USER_UNKNOWN. Looking at the PCAP, the I-CSCF sends a diameter msg to the HSS asking for the verification the public ID of: Public-Identity: sip:+alice@ims522.irisims.org The diameter msg from HSS is returning : AVP: Experimental-Result-Code(298) l=12 f=-M- val=DIAMETER_ERROR_USER_UNKNOWN (5001) Why do we want the I-CSCF to add a + to all URI's don't start with a +? The I-CSCF is the module that is adding the +, assuming that we should be able to included Kamailio.cfg for respective modules, is there a directive or variable that I need to set? Or do I simply comment out that code? Thanks, Martin Woscek The MITRE Corp. mwoscek(a)mitre.org<mailto:mwoscek@mitre.org> Phone: 703-983-2650 FAX: 703-983-7142 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Tuesday, June 11, 2019 6:22 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Woscek, Martin W. <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio 5.2.2 ICSCF changing sip to tel Hello, that prefix operation is done only if the r-uri username does not have +. So if you HSS is returning the r-uri with +, then all is ok, no other + is added. On the other hand, I am not sure what is the error or the issue you try to sort out. The error message is because of the xlog(...) in the config. Cheers, Daniel On 10.06.19 16:17, Woscek, Martin W. wrote: Hello all, The HSS is responding with 604 errors when the I-CSCF checks the public identity of the caller. PCAP attached. It looks like our I-CSCF is modifying the sip uri's to tel uri's (e.164-like), referenced attached PCAP: in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.irisims.org<mailto:alice@ims522.irisims.org> syslog output: Jun 6 16:50:06 imskamailio /usr/local/sbin/kamailio[14066]: ERROR: <script>: $ru => sip:+alice@irisims.org Looking at the default (out of the box) Kamailio.cfg for I-CSCF the following script snipet is where the modification to our sip URI is happening: # ------------------------- request routing logic ------------------- # main routing logic route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } Is there a I-CSCF setting directive we missed? Or should we simply comment the above? Any advice is appreciated. Thanks, Martin Woscek _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>

2 1

Relay NAP Probing
by Duarte Rocha 11 Jun '19

11 Jun '19

Greetings, I have a Kamailio as Sip Proxy between my clients and my Media gateway. I have NAPs (One per client) for clients and NAPs for the MediaGateway (One for each client). The MediaGateway probes Kamailio in order to see if it's alive and Kamailio probes clients. However, i would like for the MediaGateway to be aware of the client status. I thought of a few solutions : - When Kamailio probes a client and it fails, both NAPs (Client Side and MG side) could be set as inactive. - Automatically relay the MG probe to the client - Some scripting that reads the result of dispatcher.list and if it finds an inactive NAP, set the MG NAP as inactive too. Is there a better/simpler way to do this? I'd rather have everything on Kamailio configuration files instead of doing exatra scripting. Any opinions ? How can i achieve this? Best Regards, Duarte Rocha

2 1

Kamailio 5.2.2 ICSCF changing sip to tel
by Woscek, Martin W. 11 Jun '19

11 Jun '19

Hello all, The HSS is responding with 604 errors when the I-CSCF checks the public identity of the caller. PCAP attached. It looks like our I-CSCF is modifying the sip uri's to tel uri's (e.164-like), referenced attached PCAP: in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.irisims.org syslog output: Jun 6 16:50:06 imskamailio /usr/local/sbin/kamailio[14066]: ERROR: <script>: $ru => sip:+alice@irisims.org Looking at the default (out of the box) Kamailio.cfg for I-CSCF the following script snipet is where the modification to our sip URI is happening: # ------------------------- request routing logic ------------------- # main routing logic route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } Is there a I-CSCF setting directive we missed? Or should we simply comment the above? Any advice is appreciated. Thanks, Martin Woscek

2 1

Operations after relay
by Nicolas Breuer 11 Jun '19

11 Jun '19

Hello, I want to do some sql operations after a 200 reply, after relaying the sip packet Is it possible or sending the sip packet is the last operation permitted ?

3 3

Connect SIP Trunk to PBX
by Chad Kenealy 10 Jun '19

10 Jun '19

Hello, I hope you can help, We are trying to connect our SIP trunk to our new PBX system, however our SIP trunk provider requires a static IP to add to the endpoint. Our new PBX provider uses a range of IPs that are connected to a domain and the IPs change all the time. Is there a way of using Kamailio that we could get our Sip Trunk provider and PBX system to work together? Regards

2 1

Question about setbflag and isbflagset usage
by Володимир Іванець 07 Jun '19

07 Jun '19

Hello everyone! I'm not sure if I correctly understood the description of functions *isbflagset* and *isbflagset* so once again I'm asking for help :) Here is what I do: 1. Asterisk sends INVITE to Kamailio. Kamailio performs lookup of available contacts and then few checks in branch_route. I'm adding *setbflag("17", $T_branch_idx);* there. 2. Later in route[NATMANAGE] I added *if (isbflagset("17", $T_branch_idx)) {* and additional logic. *isbflagset* results in *true* (for same INVITE). 3. Finally one of UACs responds with 200 OK. Asterisk generates ACK packet that gets in route[NATMANAGE]. *isbflagset* results in *false*. So my question is: shouldn't *isbflagset* be *true* for ACK too since it's the same branch or I interpreted these fenctions incorrectly? Thanks a lot!

3 6

Re: [SR-Users] [EXT] Re: ICSCF mysql error
by Bao, Yin 07 Jun '19

07 Jun '19

Yes I think you are right it’s the mismatch of uri. However we don’t know which configuration resulted that mismatch. Here are a bit more details: ICSCF prepared this query: select * from impu where identity=? And it provided sip:+alice@irisims.org as the parameter for the query However the HSS impu table as well as the SIP INVITE only contains: sip:alice@irisims.org I am suspecting the additional + may be causing this mismatch. But where is the + coming from and why ICSCF add this in front of original INVITE line uri? Thanks! Yin From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of YASIN CANER <caner_yaso(a)hotmail.com> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Friday, June 7, 2019 at 2:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] ICSCF mysql error Hello , It is a mismatch inconming uri and your uri in hss. Yasin Caner Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> şunları yazdı (7 Haz 2019 09:28): Hello, not an expert on this particular module, but this is the internal error path from the module that probably caused this error reply: - case RC_IMS_DIAMETER_ERROR_USER_UNKNOWN, will send message with: - #define MSG_604_USER_UNKNOWN "Does not exist anywhere - HSS User Unknown" The status comes probably from the CDP module: diameter_epc_code_result.h 57: RC_EPC_DIAMETER_ERROR_USER_UNKNOWN = 5001, Maybe this helps. Henning Am 03.06.19 um 06:43 schrieb Trinh, Jenny: Hi, I’m getting the error 604 when ICSCF did a query to mysql and failed with an INVITE. Please see my attached pcap. Would appreciate any feedbacks on what might the issue. We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: SIP/2.0/UDP 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: <sip:bob@ims522.irisims.org><mailto:sip:bob@ims522.irisims.org>;tag=594048364#015#012To: <sip:alice@ims522.irisims.org><mailto:sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 req_src_ip=10.0.20.1 req_src_port=6060 in_uri=sip:alice@ims522.irisims.org<mailto:in_uri=sip:alice@ims522.irisims.org> out_uri=sip:+alice@ims522.ir<mailto:out_uri=sip:+alice@ims522.ir> Thank you _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] Using DNS in Dispatcher for SRV Clarification [SOLVED]
by JR Richardson 07 Jun '19

07 Jun '19

Hi Federico, "use_dns_cache=on" this is on be default. "use_dns_failover=on" this is off by default, changing to "on" did the trick, auto failover to next SRV server is working quickly, as expected now. Thanks for the pointer. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope > > Hi, > if you want failover support, you need to have a couple of core dns > parameters enabled: > https://www.kamailio.org/wiki/cookbooks/5.2.x/core#use_dns_cache > > and > > https://www.kamailio.org/wiki/cookbooks/5.2.x/core#use_dns_failover > > The latter defaults to off. > > Cheers, > > Federico > > On Thu, Jun 6, 2019 at 10:10 PM JR Richardson <jmr.richardson(a)gmail.com> > wrote: > > > Hey All, > > > > I'm in the lab testing kamailio 5.2 with DNS SRV records. The scenario > > is simple, using dispatcher with hard coded IP URIs > > [sip:10.10.10.10:5060] and also a DNS name URI [sip:sip.domain.com]. > > There are 2 SRV servers. > > > > The DNS SRV records are setup and resolving OK, during startup, > > kamailio list the DNS name in the dispatcher list. > > > > During a call scenario, when the dispatcher selects the DNS name URI > > for the next destination, the TM module takes over, resolves the > > _sip._udp.sip.domain.com for both SRV records OK and sends the call to > > the first SRV server in the list. But if I take the first SRV server > > offline, make another call, the TM module does not route to the second > > SRV server until the DNS record timeout is reached, in my lab was set > > for 300 seconds. I decreased the DNS SRV record TTL down to 5 seconds > > with same results above, but 5 seconds lapse when TM routes call to > > next SRV server. > > > > I think this is working as expected but not sure if this is by design > > so my question is, should the TM module route to another SRV server in > > the list in the event of a no response timeout or send the transaction > > back to the dispatcher to select another target? > > > > Should I play around with the TM Module parameter > > "max_noninv_lifetime" to get quicker results or it is better to have > > DNS SRV record TTL set really low? > > > > Thanks. > > > > JR

1 0

reading php script result with exec_avp
by Gertjan Wolzak 07 Jun '19

07 Jun '19

Hello Kamailions, I have the following challenge. I want to create a time dependent callerid, based on the time of a outgoing call a callerid will be set. I have a php script, that provides me an id number which relates to an cli. When I run that php script from the console, the id is shown in the console. When I run the script within the kamailio configuration, exec_avp("../script.php", "$avp(s:cliId)"); ,no output commande is saved in the avp variable. I think I am not presenting the result of the php script correctly. I have tried with an echo $variable; and with and fwrite(STDOUT, $variable); but no luck. I'm not a PHP coder so I am not surprised of the missing luck, but maybe somebody has a golden tip for me? Rgds, Gertjan

2 7

Re: [SR-Users] ICSCF mysql error
by Henning Westerholt 07 Jun '19

07 Jun '19

Hello, not an expert on this particular module, but this is the internal error path from the module that probably caused this error reply: - case RC_IMS_DIAMETER_ERROR_USER_UNKNOWN, will send message with: - #define MSG_604_USER_UNKNOWN "Does not exist anywhere - HSS User Unknown" The status comes probably from the CDP module: diameter_epc_code_result.h 57: RC_EPC_DIAMETER_ERROR_USER_UNKNOWN = 5001, Maybe this helps. Henning Am 03.06.19 um 06:43 schrieb Trinh, Jenny: Hi, I’m getting the error 604 when ICSCF did a query to mysql and failed with an INVITE. Please see my attached pcap. Would appreciate any feedbacks on what might the issue. We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: SIP/2.0/UDP 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: <sip:bob@ims522.irisims.org><mailto:sip:bob@ims522.irisims.org>;tag=594048364#015#012To: <sip:alice@ims522.irisims.org><mailto:sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 req_src_ip=10.0.20.1 req_src_port=6060 in_uri=sip:alice@ims522.irisims.org<mailto:in_uri=sip:alice@ims522.irisims.org> out_uri=sip:+alice@ims522.ir<mailto:out_uri=sip:+alice@ims522.ir> Thank you _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 1

Using DNS in Dispatcher for SRV Clarification
by JR Richardson 07 Jun '19

07 Jun '19

Hey All, I'm in the lab testing kamailio 5.2 with DNS SRV records. The scenario is simple, using dispatcher with hard coded IP URIs [sip:10.10.10.10:5060] and also a DNS name URI [sip:sip.domain.com]. There are 2 SRV servers. The DNS SRV records are setup and resolving OK, during startup, kamailio list the DNS name in the dispatcher list. During a call scenario, when the dispatcher selects the DNS name URI for the next destination, the TM module takes over, resolves the _sip._udp.sip.domain.com for both SRV records OK and sends the call to the first SRV server in the list. But if I take the first SRV server offline, make another call, the TM module does not route to the second SRV server until the DNS record timeout is reached, in my lab was set for 300 seconds. I decreased the DNS SRV record TTL down to 5 seconds with same results above, but 5 seconds lapse when TM routes call to next SRV server. I think this is working as expected but not sure if this is by design so my question is, should the TM module route to another SRV server in the list in the event of a no response timeout or send the transaction back to the dispatcher to select another target? Should I play around with the TM Module parameter "max_noninv_lifetime" to get quicker results or it is better to have DNS SRV record TTL set really low? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Re: [SR-Users] ICSCF mysql error
by Daniel-Constantin Mierla 06 Jun '19

06 Jun '19

Hello, check you kamailio configuration file, the 604 reply is likely to be generated there -- although I haven't written any of the ims modules, typically the reply generated by kamailio are from configuration file, not hardcoded in the source. Cheers, Daniel On 03.06.19 06:43, Trinh, Jenny wrote: > > Hi, > > I’m getting the error 604 when ICSCF did a query to mysql and failed > with an INVITE. Please see my attached pcap. Would appreciate any > feedbacks on what might the issue. > > > > We’re using BOGHE client to register and make call with Kamailio IMS 5.2.2 > > > > May 22 09:15:10 imskamailio /usr/local/sbin/kamailio[22167]: DEBUG: > <core> [core/msg_translator.c:2305]: generate_res_buf_from_sip_res(): > copied size: orig:143, new: 56, rest: 630 msg=#012SIP/2.0 604 Does not > exist anywhere - HSS User Unknown#015#012Via: SIP/2.0/UDP > 0.0.0.0:4060;received=10.0.20.1;branch=z9hG4bK3c92.7402885f6124cf2862bde6f9458080b5.0#015#012Via: > SIP/2.0/UDP > 10.141.161.85:56962;received=10.141.161.85;branch=z9hG4bK594030036;rport=56962#015#012From: > <sip:bob@ims522.irisims.org>;tag=594048364#015#012To: > <sip:alice@ims522.irisims.org>;tag=077cecf844e3c5622044f0dc84818fc7-09fb#015#012Call-ID: > 25759faf-6f50-5607-c8f9-162abbbff8cb#015#012CSeq: 19622 > INVITE#015#012Server: Kamailio I-CSCF#015#012Content-Length: > 0#015#012Warning: 392 10.0.20.1:5060 "Noisy feedback tells: pid=22149 > req_src_ip=10.0.20.1 req_src_port=6060 > in_uri=sip:alice@ims522.irisims.org out_uri=sip:+alice@ims522.ir > > > > Thank you > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Use dmq module with private and public IP addresses
by Markus Bönke 05 Jun '19

05 Jun '19

Hello NG, I have a question regarding the dmq / dmq_userloc module. In my test setup have 2 Kamailio servers in 2 different data centers. The data centers are interconnected by a crosslink. Each Kamailio server has configured an internal (private) and an external (public) IP address. The internal IPs are routed to use the crosslink. The external IPs are routed through the public internet by different ISPs. The requirement is that the replication should primarily be routed via the crosslink, only if the crosslink is down it should be routed via public internet. As I understand the „server_address“ parameter can have only one IP address. If I configure the public IPs on both sides and one ISP connection gets disconnected, packets will NOT be routed automatically via the crosslink. And the other way around, if I configure the private IPs on both sides and the crosslink gets disconnected, packets will NOT be routed automatically via public internet. Is there any way to setup and use dmq for such a scenario ? Thanks and regards Markus

2 3

IMS - change the transport layer protocol for Diameter based interfaces (especially Rx)
by "björn bauer" 05 Jun '19

05 Jun '19

3 2

RTPEngine and dialog moving.
by Igor Olhovskiy 04 Jun '19

04 Jun '19

Hi! Wanted to clarify some info about rtpengine working within HA-environment. Idea is we have multiple rtpengines under same setid. So, when I'm calling rtpengine_offer() I really don't care which is selected. But than might happen situation, where this call(dialog) is moved to other Kamailio instance (via dlg_db_load_callid()) and from there called rtpengine_delete(). So, my question is, how I can be sure, that delete request will go to same rtpengine instance that was selected in rtpengine_offer() on different server. And where this info is stored, within dialog or within TM module (which is not replicated in any way) Regards, Igor

3 4

Info: app_lua_sr module - old Lua sr library
by Daniel-Constantin Mierla 04 Jun '19

04 Jun '19

Hello, several days ago I split the app_lua module, relocating the code related to old Lua sr library to app_lua_sr module, keeping in this way app_lua smaller and focused to the KEMI code. Over the time, app_lua_sr should be deprecated and removed, as we are sure the functions in Lua sr have an equivalent in Lua KSR. Therefore, if you are using in Lua scripts functions from 'sr' library, consider migrating to KSR as soon as possible. The intermediary solution for a quick upgrade to master version (to be release as v5.3.x) is to load app_lua_sr and update some modparams, like: loadmodule "app_lua.so" loadmodule "app_lua_sr.so" modparam("app_lua_sr", "register", "sl") If you find a function from Lua sr not available in Lua KSR, then report it via mailing lists or bug tracker. An article about this topic was published at: * https://www.kamailio.org/w/2019/05/new-module-app_lua_sr/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Multiple definitions with OR or AND in "ifdef" in kamailio script
by Maxim M 04 Jun '19

04 Jun '19

Is it possible to do something like this in kamailio script?: !!ifdef DEF1 || DEF2 .... !!endif

2 1

empty action lists in route block
by Maryam Baghdadi 04 Jun '19

04 Jun '19

Hello everyone! I tried to use from Lua scripting language in kamailio configuration and rewrite all routes in Lua script. and I export many functions in ims_auth in KEMI and recompile ims_auth module. I have a problem with www_ims_challenge function calling in Lua script. when I define and call a function (REG_MAR_ROUTE) in www_ims_challenge as a route black, I get this error " empty action lists in route block" . why this happens and what's the solution ? thanks in advance

3 2

Removal of Dispatcher
by Colin, Andrew 04 Jun '19

04 Jun '19

I am using the API to remove dispatchers with a simple curl command, is there a way i can make kamailio check that there are sufficient dispatchers available before removing them when the command is run? Example: i am removing asterisk1 out of kamailio dispatcher list, but kamailio needs to ensure that there is atleast 2 or 3 other asterisk servers active in the list before it allows it to be removed. any ideas how i can achieve this?

2 3

AAR change subscription ID
by eyas barhouk 03 Jun '19

03 Jun '19

dear community, when i'm trying to deploy kamailio IMS with PCRF i get ims-qos AAR error due to Subscription id. in ims_qos module i found that i can change the subscription_id type and value as following: * subscription_id to hard code subscription ID for AAR. Used for some broken PCRFs. Leave blank to use default * subscription_id_type to hard code subscription ID type for AAR. Only applicable if subscription_id is set. Set to -1 to use default. This is as per RFC 4006: END_USER_E164 0, END_USER_IMSI 1, END_USER_SIP_URI 2, END_USER_NAI 3, END_USER_PRIVATE 4 but when i changed it there was no change on AAR Subscription-id AVP. any advise about how to set or change the subscription id dears ?? B.R Eyas

1 0

Releasing a new version for Siremis
by Daniel-Constantin Mierla 03 Jun '19

03 Jun '19

Hello, after quite a bit of time, we are considering to release a new version for Siremis - the web management interface for Kamailio. A lot of effort has been pushed to address the issues with PHPv7.x, which deprecated a lot of PHPv5.x API as well as made more strict rules in regards to classes and inheritances. At this stage it should be in good stage, but it would help if others can give it a ride to Siremis master branch: * https://github.com/asipto/siremis Installation guidelines: * https://kb.asipto.com/siremis:install51x:main Testing with both PHP v5 and v7 is appreciated in order to be sure things were not broken for old systems. Any issue found should be reported on github project issue tracker. If all goes fine, in soon we will make the release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

invalid payload location
by Kelvin Chua 31 May '19

31 May '19

is there a plan to create an sdp sanity functionality? ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location opensips has something similar sipmsg_validate() *'s'* - checks the integrity of the SDP body, if it exists Kelvin Chua

2 2

Table creation error when enabling ims_usrloc_scscf module.
by Ali Shirvani 30 May '19

30 May '19

Hi all, I installed kamailio v5.2.3 from source and IMS modules included. Then I tried to create required database and tables, I removed "dialog" from STANDARD_MODULES and added "ims_dialog ims_charging ims_usrloc_scscf ims_usrloc_pcscf presence" to the EXTRA_MODULES in kamctlrc file. When I issue `<kamailio_home>/sbin/kamctl create` I got following error: ERROR 1062 (23000) at line 39: Duplicate entry 'subscriber' for key 'table_name_idx' ERROR: Creating extra tables failed at ims_usrloc_scscf! Should I exclude an other module from STANDARD_MODUELS? Regards, Ali

2 1

turn on DEBUG level debugging for ICSCF module
by Trinh, Jenny 30 May '19

30 May '19

Hi, How do I turn on DEBUG level debugging for ICSCF module? Thanks Jenny

2 1

app_java.so: undefined symbol: JNI_CreateJavaVM please help
by Renato Gallo 30 May '19

30 May '19

gcj is not a requirement since 2018 [ https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html | https://www.codesd.com/item/undefined-symbol-jni-createjavavm-on-linux.html ] all details are in the bug ticket at the link: [ https://github.com/kamailio/kamailio/issues/1969 | https://github.com/kamailio/kamailio/issues/1969 ] please help Renato Gallo System Engineer sede legale e operativa: Via Privata Cefalonia, 14 - 20156 - Milano (MI) Tel. +39 02 - 87049490 Fax +39 02 - 48677349 Mobile. +39 342 - 6350524 Wi | FreeNumbers: https://freenumbers.way-interactive.com Wi | SMS: https://sms.way-interactive.com Wi | Voip: https://voip.way-interactive.com Asterweb: http://www.asterweb.org Le informazioni contenute in questo messaggio e negli eventuali allegati sono riservate e per uso esclusivo del destinatario. Persone diverse dallo stesso non possono copiare o distribuire il messaggio a terzi. Chiunque riceva questo messaggio per errore è pregato di distruggerlo e di informare immediatamente [ mailto:info@sigmaware.it | info@ ] asterweb.org

1 0

Fwd: Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 30 May '19

30 May '19

Hi Daniel, See pastebin url below for the backtrace. https://pastebin.com/tTUUk4YH Hopefully I did it the right way. Please let me know if you need anything else. FYI, Kamailio is running inside of a docker container. version: kamailio 5.2.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 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_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 6.3.0 I can easily replicate this, but to be sure I've saved the core file locally on my machine. Regards, Grant On Thu, May 2, 2019 at 10:05 AM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > can you get a core file for this case? > > Set `ulimit -c unlimited` before starting Kamailio as root and run again. > > Once you have the core file, get the backtrace with gdb. > > Cheers, > Daniel > > On 01.05.19 20:26, Grant Bagdasarian wrote: > > Hello, > > > > I was trying out the auth_ephemeral module inside the xhttp > > event_route in Python KEMI, and discovered a possible bug. > > > > https://pastebin.com/4Ghumfis > > > > The following will result in the stack smash error. Whenever I leave > > out the unixtimestamp the > > module will complain about not being able to convert to an int, which > > make sense, but doesn't lead to a stack smash error and a complete > > crash of Kamailio. > > > > KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). > > > > According to the documentation this module can only be used in > > REQUEST_ROUTE, but I found a presentation online which also used this > > module and that particular function in event_route, which is what I > > need for WebSocket authentication. > > > > Hope this helps. Please let me know if you need more information. > > > > Regards, > > > > Grant > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com >

2 10

Kamailio Load Balancing
by Gaurav Bmotra 30 May '19

30 May '19

HI i am using kamailio 5.1.2 With Ubuntu 18.04.2 LTS and using Freeswitch as Media server and kamailio as Register server i wan to implement load balance on this plz help thank you Gaurav -- *Regards:* Gaurav Kumar

3 7

usrloc extra attributes and db_mode=0
by Patrick Wakano 29 May '19

29 May '19

Hello list, Hope you all doing well! I am trying to use the extra attributes (xavp_contact) of the usrloc module to save some additional info about the user. I am setting the value before the save() and doing a call to registered() (not lookup()) before trying to access these extra attributes. This works fine with db_mode=3 but does not work in case of db_mode=0. Does anyone knows if it should also work with mode 0? I was expecting it to work but a call to kamcmd ul.dump show nothing related..... I don't want to use the DB for the location purposes because I prefer to have multiple servers using the dmq_usrloc (that works like a charm!) and it conflicts with DB persistence of the location table (when the DMQ replication happens, all servers (sharing the same DB) try to save the same user info in the location table) Thank you, Kind regards, Patrick Wakano

2 6

Statistics help
by Duarte Rocha 29 May '19

29 May '19

Greetings, I'm doing some monitoring of my Kamailio in order to quickly identify system failures and overloads. In order to do that i have scripts that read the output of "kamcmd sl.stats" and identify the rising of "500_replies", "5xx_replies" and "6xx_replies". However, i have rate limiting on my Kamailio, which generates "503" replies when limit is reached, which generates an unnecessary alarm. Is there a way to get more specific stats about the repleis generated by Kamailio? If that isn't possible, will i be safe only monitoring "500" and "6xx" replies? In my experience, when the server has reached a processing or memory limit, the reply is always 500, but i'm not really sure about this. Thank you for your help. Best Regards, Duarte Rocha

2 1

Push proxy and load balancer for asterisk?
by Gerrit Jacobsen 29 May '19

29 May '19

Good morning, I want to realise a push notification proxy to signal incoming calls from asterisk to softphones. Is Kamailio the preferred solution for this? I looked also at Flexisip and Asterisk scripts and was told by a seasoned professional that Kamailio is the preferable solution as its scalable. True? I would be glad if someone could share his experiences. Also can one Kamailio instance be used as an asterisk load balancer and also push notification proxy ? I have been a heavy openser user a long time ago. I used first openser as my core routing engine but then switched to yate. Please excuse my rather broad questions. From my experience with Openser the devil is in the detail. So any hints on where this setup may cause problems would be great. Thanks a lot for your input. Gerry

3 2

Kamailio Lord Balance
by Gaurav Bmotra 29 May '19

29 May '19

hi i m using kamailio 5.1 with Ubuntu 18.04.2 LTS and freeswitch as Media server -- *Regards:* Gaurav Kumar

4 4

RFC: split out old sr api from app_lua
by Daniel-Constantin Mierla 29 May '19

29 May '19

Hello, I am considering to split the app_lua module in order to move the older style API available under Lua 'sr' module in a dedicated module (like app_lua_sr). Later this module can be obsoleted and removed, once we know that everything there is covered by KEMI KSR Lua module. In this way, we make it more clear that KSR is the preferred API to use as well have the app_lua smaller, for those using only KSR and not needing 'sr' API. Therefore I am starting the discussion to see if anyone has something against or other suggestions. I cc-ed sr-dev to make developers aware, but I think the best place for discussion is sr-users, being related to how Kamailio is used, therefore reply only to sr-users. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

RTPEngine / Transcoding question
by Carsten Bock 29 May '19

29 May '19

Hi, I want to implement selective transcoding, e.g. avoiding Transcoding between a HD codec (G722) and a non HD Codec (G711). In case the caller does offer HD codecs, this is fine, e.g. if(sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_offer("codec-transcode=OPUS codec-transcode=G722 codec-transcode=PCMA"); } else { rtpengine_manage("codec-transcode=PCMA"); } Now I have the issue, if the callee only sends me G711, I don't want to offer G722 or Opus to the caller. However, rtpengine_answer() does not seem to accept "codec-strip=G722 codec-strip=OPUS", e.g.: onreply_route() { if(!sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_ manage("codec-strip=OPUS codec-strip=G722"); } } As a result, the SDP always contains the transcoding options, e.g. G722 and OPUS. I always end up in transcoding G722 to G711, which is meaningless in that case. Any ideas on how to solve that? (above examples very simplified) Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

2 5

RTPEngine: rtpengine_manage(), Subsequent requests, Transcoding
by Carsten Bock 29 May '19

29 May '19

Hi, quick question: If understood it correctly in the past (and it worked for me quite well this way), I can simply call rtpengine_manage() for subsequent requests without adding any additional information (e.g. used interfaces, RTP/SRTP translation or anything like that), as this information is stored in the internal hash table of the RTPEngine module during call setup and initial offer/answer. However, this does not seem to apply to the transcoding options. When I just call "rtpengine_manage()" for a Re-INVITE of a transcoded call (in my case AMR-WB to G722), the transcoding options are no longer applied. The forwarded offer does not get the transcoded codecs added (e.g. in my case G722 is not added to the list of AMR/AMR-WB from the phone). Did I misunderstand something or is it a bug? My RTPEngine is still on 7.2, but I will double check with more recent versions today or tomorrow. However, I also found no commit messages relating to this. Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

3 2

Call Routing at IMS
by Iman Mohammadi 29 May '19

29 May '19

Hi Everyone When P-CSCF receives INVITE from client it sends INVITE to S-CSCF then S-CSCF without any look up(checking the user is registered or not)sends INVITE to I-CSCF then I-CSCF sends LIR and receives LIA and sends INVITE to same S-CSCF (in case of one S-CSCF exists).My Question is here why without any look up S-CSCF sends INVITE to I-CSCF??this procedure affects system performance and increases call set up time

2 1

WebRTC bad message (offset: 22)
by Ilie Soltanici 28 May '19

28 May '19

Hello, Can someone help me to discard those messages from kamailio log files? ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS 212.2.161.34:60477#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: VeiXbJClWneIrIyYgM5v3XKYHLA=#015#012Server: WebRTC GW#015#012Content-Length: 0#015#012#015#012> As I understand this is just an HTTP response to the first request, and shouldn't go through config file execution, but I don't understand why it is getting through it. Thank you

1 0

kamailio webrtc 183 early media.
by Zhan Bazarov 28 May '19

28 May '19

Hello! I have setup kamailio+rtpengine+asterisk.. And I have issue with early media for WEBrtc endpoints... As I understood right, for this case I need to use rel100(PRACK) for sending early media... But I didn't find any example for solve this issue... Can you help me with it? It would be greatly appreciated!

1 0

mem leak on 5.1.2?
by Andrew Chen 27 May '19

27 May '19

Hey guys, Need some help with possible run away memory issue on my 5.1.2 Kamailio. I noticed I'm getting this memory error from kam logs: May 24 13:47:52 ashintgtpsg51 /usr/sbin/kamailio[1857]: WARNING: http_client [functions.c:298]: curL_query_url(): Curl library out of memory (url:https://<some url>) my -m on kamailio startup is currently set to 64 but there is only 1 call going through. In our check_mk, I noticed the memory usage is ramping up: [image: image.png] What's the cause of this? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

4 8

Getting the following error when sending INVITE request for calling
by Mohit Chawla 27 May '19

27 May '19

Hi , I am getting the following error in trying to send INVITE message from SIP client. {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: timestamp=1558902843;method=INVITE;from_tag=69588468;to_tag=ebd29a60;call_id=4_Do2VIMWLR6Wf5ubiBxgg..;code=200;reason=OK;src_user=1016;src_domain=172.25.22.23;src_ip=172.25.22.40;dst_ouser=1015;dst_user=1015;dst_domain=172.25.22.40 May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query May 27 02:04:03 epc-srv-03 /usr/sbin/kamailio[17238]: {2 2 INVITE 4_Do2VIMWLR6Wf5ubiBxgg..} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert into database May 27 02:04:10 epc-srv-03 /usr/sbin/kamailio[17229]: {2 3 BYE 4_Do2VIMWLR6Wf5ubiBxgg..} NOTICE: acc [acc.c:281]: acc_log_request(): ACC: transaction answered: *Can anyone help me understand the reason for this error.* *I am attaching the used kamailio.cfg file in this message.* *Thanks and Regards,* *Mohit* -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 6

remove_hf() leaves junk in the headers?
by Oded Arbel 27 May '19

27 May '19

I'm trying to rewrite the `From` header in an invite going out to an external system (*), by calling remove_hf("From"); followed by append_hf("From: <some value>\r\n"); but the result is an invalid SIP message, because apparently after `remove_hf()` finishes, it leaves the content of $fU in place of the `From` header - just the raw text, without a header name or even a "\r\n" - and that eats the next header line and breaks the INVITE. Here is how the INVITE originally looks (without either `remove_hf()` or `append_hf()`): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=649.f0b> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK6188.ff5c4fd855bd2970fdf32d4807b114c2.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK581897a4;rport=5066 Max-Forwards: 69 From: <sip:972515340014@example.com:5066>;tag=as0bcee217 To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 722299dc15aad53d28172d433e8665f4@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:42:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ad199638-4df3-4aef-81f1-1b50581032a7 Content-Type: application/sdp Content-Length: 280 v=0 o=root 817636675 817636675 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39064 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39065 a=ptime:20 ---8<--- and here is another call with the same setup, with just `remove_hf("From");` being invoked (without calling `append_hf()` - this is just for reproduction): ---8<--- INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0 Record-Route: <sip:10.90.80.70;lr;did=f45.91b1> Via: SIP/2.0/UDP 10.90.80.70:5060;branch=z9hG4bK024d.2690ebf7a809edca2208e6b2561ccf9e.0 Via: SIP/2.0/UDP 10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK79ae3073;rport=5066 Max-Forwards: 69 972515340014To: <sip:972522353283@10.20.30.40:5060> Contact: <sip:972515340014@10.90.80.71:5066> Call-ID: 7b20ba5a549f6f886c22985146765736@10.90.80.71:5066 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.7.2 Date: Sun, 26 May 2019 14:44:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces X-CID: ffbb4ac1-3a0f-4dd6-9ce9-7ad24361081a Content-Type: application/sdp Content-Length: 280 v=0 o=root 509740923 509740923 IN IP4 10.50.60.70 s=Asterisk PBX 14.7.2 c=IN IP4 10.50.60.70 t=0 0 m=audio 39102 RTP/AVP 8 0 18 9 a=maxptime:60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:39103 a=ptime:20 ---8<--- As you can see, the `From` header is indeed gone, but it left the $fU just before the `To` header. WTF? I'm running Kamailio 5.2.2 built locally. Notes: *) the problem I'm trying to solve is that I can't get my Asterisk installation to generate the `From` header I want, the reason is out of scope of this question, so I'm sending custom SIP headers over to Kamaiio and I want to rewrite the header in Kamailio to the "correct" value. -- Oded Arbel

6 8

PRE-PAY update timer
by Steve Bucklin 25 May '19

25 May '19

Hello all (again), I am looking at a 'good' method to start a call with "chunks" of time allocated. I wish the initial call to check a credit, and if credit is good, allow a small amount of time (say 30 seconds). I have enabled DIALOG and have looked at CNXCC, but am confused! When the initial 30 seconds runs out, I want to authorise another 30 seconds, etc, etc, etc. I have started to play and set a dialog timer that expires to a route, and then can 'return' allowing the call to continue - BUT was not able to reset the dialog timer to trip again after another 30 seconds. I tried to 'set_max_time' on cnxcc, but this did not seem to trip the event route upon timeout? Does anyone have a pointer to any examples? I also am unsure of how to update the timer and keep a call in play! Steve

5 4

No Private memory left using usrloc with db_redis in mode 3
by José Seabra 24 May '19

24 May '19

Hello there, i have stored ~200 000 registrations contacts in redis database and when kamailio starts looking for expired contacts in redis then it starts giving these errors: WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires' *WARNING: PR_LOG: 123: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7ff80922b010, 3055920); Free fragment not found!* *ERROR: PR_LOG: 123: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7ff80922b010, 3055920) called from core: db_res.c: db_allocate_rows(183), module: core; Free fragment not found!* *ERROR: PR_LOG: 123: <core> [db_res.c:185]: db_allocate_rows(): no private memory left* *ERROR: PR_LOG: 123: db_redis [redis_dbase.c:1115]: db_redis_perform_query(): Failed to allocate memory for rows * I have configured the private memory to 128 MB but seems that isn't enough to handle with all these contacts. Is the only option here increase the private memory or there is any other kind of configuration(in redis or kamailio) that i can make to avoid this? Best Regards José Seabra -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Kamailio IMS installation from pacakges
by Mohit Chawla 24 May '19

24 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit From: Mohit Chawla [mailto:mohit.chawla@azcom.in] Sent: Wednesday, May 22, 2019 5:57 PM To: 'sr-users(a)lists.kamailio.org' Subject: Kamailio IMS installation from pacakges Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 3

Use of kamailio as IMS server to integrate with LTE Core Network
by Mohit Chawla 24 May '19

24 May '19

Hi , We have successfully installed the kamailio IMS server on centOS 7 using the following link (http://www.kamailio.org/wiki/tutorials/ims/installation-howto) with appropriate changes for command in CentOS 7. We are currently trying to integrate the PCSCF module with LTE Core network , but I don't have any idea how to do this. Neither I am able to find any tutorials which I can use to understand how kamailio works. Can anyone share any reference document to help me understand the kamailio working in my use case. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

listen (bind) to variable ip and port
by Roiss-Weiss, Roman 23 May '19

23 May '19

Hi guys, is it somehow possible to bind to a variable ip-address and port? e.g. getting the ip and port from a database? Kind regards, Roman

3 2

Kamailio CDRs for calls interrupted without BYE
by Ildar Mukhametzhanov 23 May '19

23 May '19

Hello! I am using dialog+acc modules to write CDRs in kamailio. Everything works great except one test. During active call I switched off both UACs preventing them to send BYE. Rtpengine reported about call interruption, but kamailio thinks that this call is in-progress for a long time. What is the right way to make kamailio know about calls interrupted without BYE. Thanks. -- Sincerely, *Ildar Mukhametzhanov* Head of software development VoiceON Telecom Services GmbH im(a)voice-on.net <mailto:im@voice-on.net> t. +49 30 398 20 500-77

3 8

warning logs - DB_Redis and usrloc in mode 3
by José Seabra 22 May '19

22 May '19

Hello there, I'm testing the db_redis with usrloc module in mode 3 and i have been noticing that every time that kamailio checks if there is contacts expired then it logs the following warning messages: *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1099]: db_redis_perform_query(): performing full table scan on table 'location' while performing query* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1102]: db_redis_perform_query(): scan key 1 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1274]: db_redis_perform_delete(): performing full table scan on table 'location' while performing delete* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 0 is 'expires'* *WARNING: PR_LOG: 5a14aea50d9d9b17-75(a)172.31.1.11 <5a14aea50d9d9b17-75(a)172.31.1.11>: db_redis [redis_dbase.c:1277]: db_redis_perform_delete(): scan key 1 is 'expires'* I'm not sure what is the rational behind of this, but having these logs being written when we have log level equal 2 in production is something unnecessary, so i'm asking if it shouldn't be better we have it as debug instead of warning? or do you forecast any performance issues using usrloc module in mode 3 with db_redis? Thank you for your support Cheers José -- Cumprimentos José Seabra

2 1

Re: [SR-Users] Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 22 May '19

22 May '19

Benoit, Alex, I appreciate the feedback, confirming my assumptions. Good news is the Vendor finally responded positively and agreed to add in configuration option to route on INVITE R-URI in next SDK release, expected later this year. In the meantime I added 'uac_replace_to' function to my proxy to re-write To: header with R-URI, so far so good. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope > > The To header is a cosmetic, purely logical commentary on the intended ultimate destination of the call, and no routing should ever be done on it. The RURI of the INVITE is the appropriate vehicle for that. > > When dealing with proxies, however, one will encounter from time to time the grim reality that there are user agents which expect the To URI and the INVITE RURI to be in alignment. For that scenario, you can use ‘uac’ module’s stateful rewriting functionally (uac_replace_to()) to modify To with Kamailio in a way that displays fidelity and esteem for protocol formalities. > > — > Sent from mobile, with due apologies for brevity and errors. > > > On May 21, 2019, at 9:32 AM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > > > > Hey Folks, > > > > I could use some feedback to see if my mind is right on this topic.I'm > > in a discussion with a software Vendor (respectfully unnamed) with > > terminal software routing on To: header info. It's my contention any > > modern SIP software should use INVITE header to retrieve DID info and > > route from there. > > > > My reasoning is due to the To: header should contain the original > > intended DID, Extension, Name, AoR or whatever else the original SIP > > transaction wanted to contact, but being in a multi-carrier, > > multi-hop, call forwarding environment, the INVITE header contains the > > hop-to-hop true intention of where the call should route at any given > > transaction. So in the case of terminal software the To: header could > > likely be irrelevant where as the INVITE header is accurate. > > > > My most recent response from the Vendor suggest I put a SIP Proxy in > > front of the terminal software and transform the To: header to > > whatever I need for this application. Although I can do this, I'm > > concerned with breaking RFC for CANCEL and BYE transaction sent back > > from the terminal software that may not be recognized by upstream > > origination carriers. > > > > Plus modifying To: header is widely frowned upon by most folks. > > > > Thanks. > > > > JR

1 0

Kamailio v5.2.3 Released
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, Kamailio SIP Server v5.2.3 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.3. For more details about version 5.2.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/05/kamailio-v5-2-3-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.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio IMS installation from pacakges
by Mohit Chawla 22 May '19

22 May '19

Hi, My name is Mohit Chawla working on LTE Network integration with Kamailio IMS server. We are facing issues in installation & Configuring of kamailio IMS on CentOS 7 due to no documentation is available on web (or at least I am not able to find one). Primarily I am interested in installing packages (not building from source code as of now) and use it for further testing. I can go with building the package from source if its more easier and quick. Can you guys help me share any document/wiki which is helpful for my cause ? I will be really grateful if someone can reply me by today. Thanks and Regards, Mohit -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

Releasing Kamailio v5.2.3
by Daniel-Constantin Mierla 22 May '19

22 May '19

Hello, I am considering to release Kamailo v5.2.3 out of latest git branch 5.2 next week, likely on Wednesday, May 22. As usual, if you are aware of issues not reported to the github bug tracker, fill a report there as soon as possible to give a chance to be analyzed and fixed. Also, if there was a fix for an issue that you were reporting, check to see if it was backported to be sure it is not forgotten. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

remove_hf inside of onreply_route
by Володимир Іванець 21 May '19

21 May '19

Hello! I was wondering if someone is using remove_hf in onreply_route and if yes on which Kamailio version? I'm using Kamailio v5.2.1 and as soon as I add remove_hf function to onreply_route Kamailio will fail to start with following error messages: ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fc1dc044010, 27120); Free fragment not found! ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fc1dc044010, 27120) called from core: core/io_wait.c: init_io_wait(524), module: core; Free fragment not found! Thank you for any suggestions!

2 2

Modern SIP Trunking, Route on To: or INVITE Headers?
by JR Richardson 21 May '19

21 May '19

Hey Folks, I could use some feedback to see if my mind is right on this topic.I'm in a discussion with a software Vendor (respectfully unnamed) with terminal software routing on To: header info. It's my contention any modern SIP software should use INVITE header to retrieve DID info and route from there. My reasoning is due to the To: header should contain the original intended DID, Extension, Name, AoR or whatever else the original SIP transaction wanted to contact, but being in a multi-carrier, multi-hop, call forwarding environment, the INVITE header contains the hop-to-hop true intention of where the call should route at any given transaction. So in the case of terminal software the To: header could likely be irrelevant where as the INVITE header is accurate. My most recent response from the Vendor suggest I put a SIP Proxy in front of the terminal software and transform the To: header to whatever I need for this application. Although I can do this, I'm concerned with breaking RFC for CANCEL and BYE transaction sent back from the terminal software that may not be recognized by upstream origination carriers. Plus modifying To: header is widely frowned upon by most folks. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

4 3

$snd(proto) mapping?
by Andrew Chen 21 May '19

21 May '19

Hey guys, Is there a numeric mapping for $snd(proto)? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

How to get started (configuring an interconnection)
by Benoit Panizzon 21 May '19

21 May '19

Dear Kamailio Users I might need a bit help with starting. We are a TSP and at the moment we use a commercial carrier grade voice switch, which is very VERY inflexible and where bugs we find either don't get fixed, or only with a large delay. As everywhere else we embrace opensource and enjoy the possibility to quickly fix bugs ourselves (we have a software development department) we are evaluating Kamailio. Basically, if we have a centralized SIP routing engine, which can record CDRS for Billing and do some Number Translations (lookup on external service or database) for (location based emergency numbers, call forwarding services, ported numbers, call spam blacklisting) we could do way more than our actual $$$ carrier switch is capable of. Kamailio sounds like perfect for this task. First step: Get a machine up and running with Kamailio and Siremis, done, that was easy. Next: Connect his SIP wise via what I would call a 'SIP carrier interconnecton', ip based authenticated trunk between our carrier switch and Kamailio so we can route some number ranges from our carrier switch to Kamailio and then continue to test with subscribers, subscriber trunks and all else we need on Kamailio. I'm quite fluent in Asterisk (our Voicemail and Announcement Services are based on Asterisk) and SIP and have come across many different PBX, but after reading parts of the documentation, and trying myself to understand how Kamailio talks to other SIP endpoints I am a bit at a loss. I would have expected that I would configure our carrier switch as 'remote SIP gateway' or whatever it would be called. But I start fearing, Kamilio works completely differently. So I might need a bit of pointing the right direction to get started. I did try with the 'CarrierRoute Management' or 'Dispatcher' functions, but I only find very little documentation on this. So how do I get started? How do I, for starters, tell Kamailio what the Hostname or IP of my Carrier switch is and how do I set a 'default' call route to this? Kind regards -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Storing an integer value into AVP from REDIS string
by Stephen Bucklin 21 May '19

21 May '19

Hi, Any quick thoughts. I am using a REDIS "GET" to retrieve a JSON obj as a string from REDIS and then the JANSSON get value to extract a key value from the string and pop it into an AVP ($avp(val)) and wish to ensure the value is an integer in the AVP. New to this, be kind :-) Steve

4 6

How to define IP Trunk in kamailio
by Mary Honey 21 May '19

21 May '19

Hello everyone, I install and setup an IMS Network by kamailio-5.2.2 . I have p-cscf, i-cscf and s-cscf in three different systems . my Hss is Fokus. Registration and call between subscribers of network work successfully . now I want to define IP Trunk in my network which can pass external calls through my IMS network. I don't know how to define IP Trunk and changes in which element(pcscf, icscf or scscf) should be done.

2 3

Rewrite contact header
by R, Rajkumar (Raj) 21 May '19

21 May '19

Hello, We are using alias routing from NATHelper module and I've a case where the alias part 'alias=ip~port~transport' needs to be removed from the contact header in 2xx message. There's contact header reference "$ct" pseudo variable but it is read only. Is there any approach available to achieve this as I haven't found anything in the forum archive search so far. thanks, raj

2 2

Unable to CDRs on "Busy Here 486" from remote side and save start and end time on successful call in table
by Shahid Mehmood 21 May '19

21 May '19

Hello, i want to save the status each call into the database. but failed to 1. record CDRs for "Busy Here 486/ service unavailable 503 " from remote side in database. 2. on successful call unable to record total time of call in database. Version of Kamailio Is *version: kamailio 4.4.7 (x86_64/linux) 35d8abflags: 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_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: 35d8ab* *Also Attached Config File* *Thanks*

1 0

kamailio Early IMS security
by eyas barhouk 20 May '19

20 May '19

hello community Does kamailio support Early IMS security authentication schema to be able to integrate with Huawei HSS ?? and if it's does how to configure it? B.R Eyas

1 0

Plans for next major release - v5.3.0
by Daniel-Constantin Mierla 20 May '19

20 May '19

Hello, a while ago, during the last IRC devel meeting, it was proposed to either do the next major release (5.3.0) before the usual summer holidays or leave it for autumn. Given that time is approaching, we have to decide. If it is going to happen before the summer holidays, probably we should freeze the development in 2-3 weeks, then do 3-4 weeks of testing and release during the first part of July. It would be a shorter development cycle, so I do not expect much needs for testing. Only a few new modules and plenty of work to consolidate and enhance the existing code. Any preference for one of the two options? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 2

Kamailio usrloc mode 3 using mongodb
by José Seabra 20 May '19

20 May '19

Hello there, I'm testing usrloc module with mongodb and i have a couple of questions to do. What i would like to know/have confirmation is if the fields that are specified in the following link https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-location-… can be used to create indexes in mongodb? I'm noticing that when a register is updated/deleted or expires kamailio prints the following INFO logs: *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (methods) type id (10), use DB1_STRING as default* *db_mongodb [mongodb_dbase.c:526]: db_mongodb_get_columns(): unhandled data type column (instance) type id (10), use DB1_STRING as default* These fields (*methods and **instance*) are stored in mongodb document with a null string, e.g: *{* * "_id" : ObjectId("5cdc775a6472740055289a11"),* * "username" : "x1000",* * "contact" : "sip:x1000@192.168.1.1:5674;transport=UDP",* * "expires" : ISODate("2019-05-15T20:33:26Z"),* * "q" : -1,* * "callid" : "sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61 <sipp-ci2-edge-sip-proxy1-20190515203220-1-21(a)172.31.2.61>",* * "cseq" : 2,* * "flags" : 0,* * "cflags" : 0,* * "user_agent" : "sipp",* * "received" : "sip:191.30.10.23:8888 <http://191.30.10.23:8888>",* * "path" : "<sip:172.31.2.61:5060;received=sip:191.30.10.23:8888;lr>",* * "socket" : "udp:172.31.2.11:5060 <http://172.31.2.11:5060>",* * "methods" : null,* * "last_modified" : ISODate("2019-05-15T20:32:26Z"),* * "ruid" : "uloc-15-5cdc7756-55-1",* * "instance" : null,* * "reg_id" : 0,* * "server_id" : 21,* * "connection_id" : -1,* * "keepalive" : 0,* * "partition" : 0* *}* Should i be concerned about that log msg? how can i avoid it? Thanks Regards -- Cumprimentos José Seabra

2 5

BYE relays Digest and others in Proxy-Authorization header
by Sergiu Pojoga 19 May '19

19 May '19

Hi ppl, Noticed BYE requests relayed from caller to callee have the *Proxy-Authorization* header including *Digest*, *nonce* and *response*. Is that by SIP definition or am I doing something wrong? My concern is the callee gets to see caller's authentication credentials. Thanks.

4 6

Centos7 Packages
by Matt Lamer 18 May '19

18 May '19

I am trying to install the kam server on centos7. When running the install I get the following: Install 5 Packages (+2 Dependent packages) Total size: 14 M Total download size: 14 M Installed size: 63 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-debuginfo-5.2.2-7.el7 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Anyone else having this issue?

2 1

"If you're struggling with your Tele callers to improve sales? you're not alone.
by Prakash 17 May '19

17 May '19

http://www.doocti.com +919790776655 Doocti Next Generation CTI for Smarter Sales Teams http://www.doocti.com Dear Customer, Greeting from Tevatel. Hope this mail finds you well. We are happy to announce the launch of our new product Doocti - Call center connector for crm What is Doocti? Doocti is a Computer telephony integration platform that can integrate your communication devices with your Helpdesk, CRM, and Other Key Tools. Increase agent productivity, improve customer experience and enable personalized service through deep customer insights at the point of interaction. Features of Doocti Single click Dial On-Screen Customer Profile Build on the Customer Profile Efficient Call Disposition Auto Dial Sticky Agent Feedback from the clients Reduce call time by 10-20 seconds per call Instantly retrieves and screen-pops customer related information Provides a better and more immediate response to the customers Offers a more personalized experience to the caller Dials out automatically and faster with a single mouse click Handles higher call volumes Increases first call resolution Intergrated with Currently we have associated with more than 20 brands. Curious to see what's under the hood? Enjoy a detailed look at the Doocti experience in a demo guided by a member of our team. Get a meeting on the calendar today. mailto:prakash.n@tevatel.com?subject=Need More Information about Doocti Request for a Demo [ mailto:prakash.n@tevatel.com?subject=Need More Information about Doocti ] Communications Simplified Contact Us: +919790776655 | prakash.n(a)tevatel.com No:76, Swamibakthan Street, Nammalwarpet, Chennai - 600012. www.tevatel.com | www.doocti.com http://viewstripo.email/?utm_source=templates&utm_medium=email&utm_campaign… ---------------------------------------------------------------------- This email was sent by prakash.n(a)tevatel.com to sr-users(a)lists.sip-router.org Not interested?Unsubscribe - https://tihk.maillist-manage.com/ua/optout?od=27218d28c96aa859ed0ac1287be0e… Update profile - https://tihk.maillist-manage.com/ua/upc?upd=164d3589027540e5&r=164d35890275… Asterfone technologies pvt ltd | Swamibakthan St, Thiruvalluvar Nagar, Nammalwarpet, Ayanavaram, Chennai, Tamil Nadu 600010 Our Privacy Policy [ ] and Terms of Use. [ ]

2 1

rtpengine sessions on timeout not freeing up RTP ports
by Istvan Mogyorosi 17 May '19

17 May '19

Hello Another rtpengine related question after having successfully implemented a dispatcher configuration. Call flow is ok, everything works well until I tried to increase call volume. Kamailio log showing "proxy replied with error: Ran out of ports" In the beginning both channels are up, then I can see a teardown on the incoming channel loosing the ability to collect DTMF. Using the CLI tool I have statistics showing that none (!) of the calls are "regular terminated", they are on timeout. After experimenting with handling BYE, I ended up could getting a "regular terminated" call by using route[RELAY] .... if (is_method("BYE")) { rtpengine_delete(); } and then I can see some good things! Received command 'delete' from 127.0.0.1:58538 Scheduling deletion of call branch 'as65b95845' (via-branch '') in 5 seconds Replying to 'delete' from 127.0.0.1:58538 (elapsed time 0.000216 sec) Please correct me if I'm doing wrong this way! cheers Istvan

1 0

RFC: Kamailio command line parameters
by Daniel-Constantin Mierla 17 May '19

17 May '19

Hello, a while ago I added support for long format parameter names for Kamailio command line. That's opening the door for more options to be set via command line arguments and I discovered that especially in a containerized environment, it is a trend to start applications using more command line parameters, than relying on config files. The current command line parameters are listed by: kamailio -h Use the master branch for it, because has many more than 5.2.x. Anyhow, the purpose of this message is to see if anyone else considers useful to add other command line parameters, and if yes, which ones. A typical candidate will be a global parameter that you have to change when relocating to/spinning up another Kamailio environment and you need to change the kamailio.cfg. Those that generate the interest of many people, can be easily added now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Testing the KEMI interpreters for functions with 4 and 5 parameters
by Daniel-Constantin Mierla 17 May '19

17 May '19

Hello, because there were many variants not handled for KEMI functions with 4 and 5 parameters, I made a script to generate the code in order to be sure none is missing. The script was pushed in Kamailio repo at: * https://github.com/kamailio/kamailio/tree/master/misc/tools/kemi You don't have to use it, I listed it for reference, just in case the code needs adjustments in the future. If anyone is using a KEMI interpreter and has functions exported by Kamailio with 4 or 5 parameters, it would be appreciated if you can run some tests with master branch and report the results. All kemi interpreter modules were updated, so it doesn't matter which one you test. If the results are good, then the patches will be backported to branch 5.2, so there will be no case of unhanded functions with these numbers of parameters. The cases with up to 3 parameters were not affected. For 6 params, still only string values are handled. The other variants can be added, but I think they are not needed for the moment, so I tried to keep it small for the moment. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio World thanks
by Daniel Pocock 16 May '19

16 May '19

Hi Daniel and Kamailio team, Well done on another successful Kamailio World in 2019. As usual, it was a great opportunity to meet people and learn new things. Regards, Daniel

2 1

siputils add_uri_param not available in KEMI python3
by Patrick Murphy 15 May '19

15 May '19

Hello, I am using the following version of Kamailio and using app_python3 with KEMI but I cannot find any exported function of add_uri_param from the siputils module. Any idea how I can perform: KSR.siputils.add_uri_param("name=value") ? version: kamailio 5.2.2 (x86_64/linux) 9260a2 Cheers, Pat.

1 1

Using a single virtual domain in a multi-domain setup
by Rhys Hanrahan 15 May '19

15 May '19

Hi All, I am trying to setup Kamailio to allow me to have multiple different domains (and IPs) treated as one. I am using the domain module for this. As an example I have: * A handset registered to Kamailio by IP address. * A SIP trunk between Asterisk and Kamilio using an FQDN that uses SRV records for redundancy on Asterisk's side. I have added all domains and IPs under the same "domain identifier". It seems the "old" domain module now uid_domain had this capability of a single virtual domain, but is this possible with the more current domain.so module? From what I've said the uid_* modules older modules from the SIP Router project, and should be replaced by more recent Kamailio modules? - https://www.kamailio.org/docs/modules/5.0.x/modules/uid_domain.html VS. - https://kamailio.org/docs/modules/5.1.x/modules/domain.html MariaDB [kamailio]> select * from domain; +----+------------------------------------+------------+---------------------+ | id | domain | did | last_modified | +----+------------------------------------+------------+---------------------+ | 1 | 12.34.56.53 | vocalpoint | 2000-01-01 00:00:01 | | 2 | 23.34.56.54 | vocalpoint | 2000-01-01 00:00:01 | | 3 | sbc.foo | vocalpoint | 2000-01-01 00:00:01 | | 4 | sbc.nsw.foo | vocalpoint | 2000-01-01 00:00:01 | | 5 | sbc.vic.foo | vocalpoint | 2000-01-01 00:00:01 | | 6 | sbc.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 7 | sbc.nsw.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 8 | sbc.vic.foo2 | vocalpoint | 2000-01-01 00:00:01 | +----+------------------------------------+------------+---------------------+ 8 rows in set (0.000 sec) I managed to get registrations working by hardcoding a static realm in the config, but when when usrloc does a lookup on a handset, Kamailio can't find user(a)sbc.nsw.foo and Kamailio returns 404. It looks like this is because in the locations table it is listed with a domain of 12.34.56.53 (because the handset is registered by IP). How should this work normally? Or is it even possible? Thanks, Rhys.

2 4

Sip proxy for Asterisk with RFC3966 and 100Rel
by paolo.visnoviz＠vipcomputers.it 15 May '19

15 May '19

Dear Sirs, a telco provider give us a trunk that require 100rel and TEL RFC 3966. Usually we use Asterisk. Now Asterisk 16 with PJSIP support 100Rel, but not RFC3966, and they don't actually works to implement it. Asterisk chan_sip, instead, support RFC3966, but not 100Rel and is not actively developed on by Digium or Sangoma. So I'm stuck. Now, do you think it was possible to use Kamailio to act like a proxy sip but implementing RFC3966 for incoming and outgoing calls and RFC3966 and 100Rel for outgoing calls? And it is possible to connect all to an Asterisk server? Thank you in advance Best regards. Paolo Visnoviz

3 9

Possible memory leak in siptrace module in kamailio 5.1/5.2
by davy van de moere 15 May '19

15 May '19

Dear, I'm using siptrace within a python handler in the KEMI framework. KSR.siptrace.sip_trace_dst("sip:<myhomerip>:9060") I notice an ever increasing usage of memory, until the system runs out of memory. (definition of a memory leak :) ). Looking at the memory information kamailio is giving me I notice this: May 14 05:49:48 server kamailio[36452]: NOTICE: PYT {INVITE} {1769595029-1513210001-1955500877} {1} : qm_sums: count= 79286 size= 5812048 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) May 14 05:51:17 server kamailio[36452]: NOTICE: PYT {REGISTER} { 4_1419955802(a)192.168.0.153} {2582} : qm_sums: count= 79335 size= 5814840 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) I notice this on kamailio 5.1.8 (from a debian repo), and on kamailio 5.2.2 (from source). I can reproduce the problem very easily, and I can apply patches in the source to try out things. Could I be wrongly using siptrace? Can I give you more debugging info? Kind regards, Davy

3 4

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