- sr-users - kamailio.org

Default memory
by Duarte Rocha 20 Dec '18

20 Dec '18

Greetings, How can i ibcrease permanently the Kamailio default private and shared memory size? Thanks in advance, Duarte Rocha

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Memory allocation
by Duarte Rocha 19 Dec '18

19 Dec '18

Greetings, I have some things that i want to udnerstand about Kamailio's memory allocation. My settings are 1024Mb for shared memory and 12Mb for private memory. I'm running Kamailio on a machine with CentOS7 and 2Gb of RAM. As i understand, at startup Kamailio will allocate 1024+12*number_of_processes megabytes of memory. Is this correct? When i set up Kamailio to run with 150 processes, after starting, i still have 500Mb of free RAM and therefore not allocated to any process. That goes against what i wrote earlier. What is the best way to plan for memory specs in my machines according to the desired number of processes and shared\private memory defined? Best Regards, Duarte Rocha

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reg_fetch_contacts always returns -1
by Denys Pozniak 19 Dec '18

19 Dec '18

Hello! I want to get some parameters from user location database via function reg_fetch_contacts(). But it returns -1 and $(ulc(caller=>aor)) is <null> respectively. Please help me to understand where issue is. Script: ... $var(c_uri)="sip:+1234567890@10.10.10.10:5065"; reg_fetch_contacts("location","$var(c_uri)","caller"); $var(rc) = $rc; xlog("L_WARNING","[LOCATION] caller=>aor: rc=$var(rc) aor=$(ulc(caller=>aor)) \n"); ... [root@kamailio]# kamcmd ul.lookup location customer1 { AoR: customer1 Contacts: { Contact: { Address: sip:+1234567890@10.10.10.10:5065 Expires: 861 Q: -1.000000 Call-ID: 00DB5F27-0B02-E966-92C3-A231C378F993(a)10.10.10.10 CSeq: 1787 User-Agent: SIPPER for PhonerLite Received: sip:2.2.2.2:15043 Path: [not set] State: CS_SYNC Flags: 0 CFlags: 64 Socket: udp:3.3.3.3:5060 Methods: 8095 Ruid: uloc-1-5c1a3512-66a6-1 Instance: <urn:uuid:00FB3C49-24E8-E811-ACAA-D72FCD766436> Reg-Id: 0 Server-Id: 1 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1545232794 Last-Modified: 1545232794 } } } [root@kamailio]# kamailio -v version: kamailio 5.1.6 (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 on 13:53:57 Nov 12 2018 with gcc 4.8.5 -- BR, Denys Pozniak

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RFC: updates to some core functions
by Daniel-Constantin Mierla 19 Dec '18

19 Dec '18

Hello, it was brought into discussions several times in the past about core functions not accepting variables in the parameters. I think it is time to update them during the 5.3 release development. For few of them, I added in the past some alternative function in the corex module (e.g., force_send_socket() in core and set_send_socket() in corex module). So, I see two options: 1) add a function with similar name in corex module and same behaviour like the one from core 2) remove the function export from the core and export one with the same name from the corex module First one will ensure that configs using the functions right now keep working without any update. The second one will be better in long term from the point of documentation (no duplicated docs), but there might be few cases that would require updates in the config -- iirc, there are some functions that can get special tokens in the parameters (like forward(uri:host, uri:port)), they will get an equivalent with variables, but old config will not be compatible. Obviously the reason for this email is to ask the developers and users what would be the preferred way from own point of view. Cheers, Daniel -- 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 Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

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passing $avp to t_relay functions.
by YAS0 CANER 19 Dec '18

19 Dec '18

Hello , is there a way to pass $avp values to t_relay function. i tried ; t_relay t_relay_udp/tcp t_relay_to t_relay_avp functions log 0(75155) ERROR: tm [tm.c:2456]: fixup_t_relay_to(): fixing ($avp(dd), 1) 0(75155) ERROR: tm [tm.c:2459]: fixup_t_relay_to(): param type 8 0(75155) ERROR: tm [tm.c:2456]: fixup_t_relay_to(): fixing ($avp(flags), 2) 0(75155) ERROR: tm [tm.c:2459]: fixup_t_relay_to(): param type 11 0(75155) CRITICAL: tm [tm.c:2549]: fixup_t_relay_to(): invalid flags <$avp(flags)> 0(75155) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:489 0(75155) ERROR: <core> [core/rvalue.c:3860]: fix_rval_expr(): failure in cfg at line: 489 col: 42 0(75155) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:494 0(75155) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:495 in addition i have a look code , in fixup functions try to resolve host values so parameteres cannot find $avp . in conclusion , does kamailio need new function to pass $avp value? Can i try to code? Thanks.

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Kazoo module problem.
by Володимир Іванець 18 Dec '18

18 Dec '18

Hello! I'm trying to test kazoo module but unfortunately it is not loading: *0(25107) ERROR: <core> [core/sr_module.c:498]: load_module(): could not open module </usr/lib64/kamailio/modules/kazoo.so>: /usr/lib64/kamailio/modules/kazoo.so: undefined symbol: event_set* It was built today together with other modules and Kamailio itself today from branch 5.2 on CentOS 6. librabbitmq version is 0.5.2-1. Does anyone has it running and if so what version is it? Thanks a lot!

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Access to tables like location, imc_rooms and imc_members when kamailio is running
by hamidreza ramezani 18 Dec '18

18 Dec '18

Hello everyone I have a question regarding kamailio database, I see when kamailio is stopped, usrloc -> location and imc -> imc_rooms module tables shows some information. but when kamailio is running it's not. is it possible to manually insert to these tables, I don't want to use IMC commands like #join #accept ... is it possible to access location table and see which users are registered when kamailio is running? Best regards Hamid

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Routing of Audio and Video Stream.
by Abhijit Dutta 18 Dec '18

18 Dec '18

Hi All, Is it possible to redirect only the audio stream towards Asterisk from Kamailio (Proxy) via RTPEngine and not to send the video stream to Asterisk. I am yet to get the complete reason why my client wants this to happen though. Thanks Abhijit

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Re: [SR-Users] Setup dmq_usrloc
by Gertjan Wolzak 17 Dec '18

17 Dec '18

Hello Kamailions, Reading and understanding is a skill in it self.. Learned a bit more of that this morning. To be able to use the dmq_ursloc, just use: if(is_method("KDMQ")) { dmq_handle_message(); } So, I have registrar replication working, on with reading to get the rest working. With regards, Gertjan

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Setup dmq_usrloc
by Gertjan Wolzak 17 Dec '18

17 Dec '18

Hello Kamailions, Im trying to setup registration replication with the use of the dmq and dmq_usrlog modules. I have configured two servers, configured kamailio on both. When I register a sip client on the one I get a nice KDMQ message on the second server, but the second server reacts with a 404 not found: U 2018/12/17 05:08:38.974495 10.10.0.1:6000 -> 10.10.0.2:6000 #3 KDMQ sip:usrloc@10.10.0.2:6000 SIP/2.0. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0. To: <sip:usrloc@10.10.0.2:6000>. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Content-Length: 360. User-Agent: kamailio (5.1.6 (x86_64/linux)). Max-Forwards: 1. Content-Type: application/json. . {"action":2,"aor":"test01(a)10.88.77.172","ruid":"uloc-5c177363-ec4-1","c":"sip:test01@10.88.77.140:59293;ob","received":"","path":"","callid":"zjucvmzUT2hNJCIfqeUXEsOaEzyHCs-4","user_agent":"CSipSimple_G3121-26/r2457","instance":"","expires":1545041786,"cseq":24435,"flags":0,"cflags":0,"q":-1,"last_modified":1545040886,"methods":8159,"reg_id":0,"server_id":0} U 2018/12/17 05:08:38.979062 10.10.0.2:6000 -> 10.10.0.1:6000 #4 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0;rport=6000. To: <sip:usrloc@10.10.0.2:6000>;tag=02f7a7b91eadf5e07797231cb2f64326-b794. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Server: kamailio (5.1.6 (x86_64/linux)). Content-Length: 0. . I think I have to configure a bit more than just the parameters of the dmq and dqm_usrloc modules. Ive googled, looked in the Sip routing with kamailio book, but no luck. Did not see any examples in the kamailio source, with regards to the dmq_usrloc module. I am not looking for direct answers, have not provided you with enough information for that, but I would appreciate information on how to correctly configure registration replication using the dmq/dmq_usrloc module... Thank you. With regards, Gertjan

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Reminding again about private/direct emails from mailing lists discussions
by Daniel-Constantin Mierla 17 Dec '18

17 Dec '18

I see that still a lot of emails are sent directly to me after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon), I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended and there is practically 0 chance to get an answer from myself. The rule is also suggested in the mailing lists presentation page: - https://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get a reply instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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P-CSCF without Rx interface
by Tsvetan Filev 17 Dec '18

17 Dec '18

Hi All. May be this has been asked before but I was not able to find an answer. I setup IMS with 3 instances of kamailio: P-CSCF, I-CSCF and S-CSCF (I got the config files from here https://github.com/kamailio/kamailio/tree/master/misc/examples/ims). I would like to register two clients to the IMS and make a VoLTE like call. So I want to setup my P-CSCF to send SIP messages to my I-CSCF without Rx interface (without using PCRF). I'm not able to find how to tell P-CSCF where to direct the sip traffic. It obviously should happen in route[REGISTER]. Right now P-CSCF sends all REGISTER requests to itself and goes to a loop. Regards.

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Sending to an announcement server from a failure_route
by Boden Garman 17 Dec '18

17 Dec '18

Hi list, I'm trying to play an announcement for my users when my PSTN gateway responds with a 404 Not Found (Reason: Q.850;cause=3), indicating a wrong number. I'm a bit lost and would appreciate any advice. What I am seeing is two parallel SIP invites, one to my announcement server (hooray!) and one to the PSTN gateway again (which I don't want). How do I stop the original branch from starting again? I've tried changing the order of rewritehostport and append_branch without any difference. I've also tried to use t_relay() instead of the RELAY route. I'm using Kamailio 5.1 and have modified the deafult kamailio.cfg Thanks! Boden I think this is the relevant parts of the config: # PSTN GW routing route[PSTN] { [PSTN regex logic omitted] # classify as an outbound call set_dlg_profile("outbound_calls"); # select a PSTN gateway from the dispatcher if (!ds_select_dst("66", "8")) { send_reply("404", "No destination"); exit; } # failover if the selected PSTN gateway is down t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destinations in failure route failure_route[RTF_DISPATCH] { xlog("inside RTF_DISPATCH"); if (t_is_canceled()) { exit; } # wrong number from PSTN if (t_check_status("404")) { append_branch(); append_hf("X-Application: announcement\r\n"); append_hf("X-Announcement-Playback: wrong-number\r\n"); route(TO_ANNOUNCEMENT); exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { xlog("selecting next destination"); if (ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Routing to announcement server route[TO_ANNOUNCEMENT] { if (!is_method("INVITE")) { return; } rewritehostport('ANNOUNCEMENT_SERVER:5060'); route(RELAY); exit; }

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Problem with selection Transport Protocol - MultiHomed Problem
by YAS0 CANER 17 Dec '18

17 Dec '18

Hello , i have trouble about selection transport protocol. i have tried so many thing but couldnt find TM selected which IPv? and protocol type (TCP or UDP) For example; i have 2 interface ipv6 and ipv4. i will route my message (INVITE-Register etc.) another node that use TCP and IPV6 , domain name is test1.test. TM module or some module resolve this domain to IPV6:UDP but i want to sent it over IPV6:TCP socket. so i cannot know which protocol is resolved and cannot route the right socket. $du variable show only sip:ali@test1.test Conclusion : There isn't any pseudo value about which destination of IP version will use. Thanks Flow: UAC (IPv4:UDP) ----------> Kamailio (MultiHomed) ----> (TCP:IPv6)Kamailio ---- Kamailo (multiHomed ) cfg ------ auto_bind_ipv6=1 dns_try_ipv6=yes dns_retr_time=1 dns_retr_no=1 dns_cache_flags=4 mhomed=1 fork=yes tcp_source_ipv6 = [fd61:c23c:b3b6::5] listen=tcp:[fd61:c23c:b3b2::5]:5060 listen=udp:[fd61:c23c:b3b2::5]:5060 listen=udp:192.168.1.107:5060 listen=tcp:192.168.1.107:5060 route []{ if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 1\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } else if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 2\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else if($dP=="UDP"){ xlog("L_INFO","[route] [UDP] 1\n"); if (!t_relay_to_udp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else{ xlog("L_INFO","[route] [??] 1\n"); if (!t_relay()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } }

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Re: [SR-Users] sr-users Digest, Vol 163, Issue 15
by Toufic Mobarak 17 Dec '18

17 Dec '18

Tnhxtx Regards, Toufic Mobarak > On Dec 15, 2018, at 6:00 AM, sr-users-request(a)lists.kamailio.org wrote: > > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Routing of Audio and Video Stream. (Daniel-Constantin Mierla) > 2. Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 3. Re: Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 4. Re: Routing of Audio and Video Stream. (Sergiu Pojoga) > 5. Re: Problem with selection Transport Protocol - MultiHomed > Problem (Daniel-Constantin Mierla) > 6. Issue on kamailio 4.1.9 of database reconnection (Emmanuel BUU) > 7. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > 8. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 14 Dec 2018 12:54:44 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, Abhijit Dutta <duttaa(a)hotmail.com> > Subject: Re: [SR-Users] Routing of Audio and Video Stream. > Message-ID: <26c51f12-5242-4794-5124-f37bc523c7ae(a)gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Hello, > > > I guess you still need to send the INVITE to Asterisk, right? > > > Cheers, > Daniel > > >> On 14.12.18 11:15, Abhijit Dutta wrote: >> Hi All, >> >> Is it possible to redirect only the audio stream towards Asterisk from >> Kamailio (Proxy) via RTPEngine and not to send the video stream to >> Asterisk. I am yet to get the complete reason why my client wants this >> to happen though. >> >> Thanks >> Abhijit >> >> _______________________________________________ >> 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 > Kamailio Advanced Training -- www.asipto.com > >

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Automating installation
by John Tuxies 14 Dec '18

14 Dec '18

Hi all. I am new to this and i would like some help please. I am looking to deploy Kamailio and Asterisk in my systems and i would like to automate the process. I have found a bash script but it it outdated from here: https://github.com/gurumelo/asteriskkamailio i did some changes and modifications to adapt it to newer version of software, but it fails to start Kamailio and continue to Asterisk. I would appreciate if someone could help me with this script and make it work. The systems i am planning to execute it is in Debian 9 and Ubuntu 18.04 server, both 64bit. Sincerely yous, John

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FreeBSD Kamailio pkg
by Sergiu Pojoga 14 Dec '18

14 Dec '18

Did anyone notice that when installing Kamailio from FreeBSD package, by default, it accepts registration with whatever account credentials? Not sure if it's like that with other OS packages (usually I install it from source), but this behavior kind of doesn't go with the popular saying that Kamailio by default "does nothing", lol. Regards, --Sergiu

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Kamailio ERROR: _reply_light(): can't generate xxx reply when a final yyy was sent out
by Olli Attila 13 Dec '18

13 Dec '18

Hello, I'm getting these error messages on kamailio log from time to time: Dec 11 19:18:30 /usr/sbin/kamailio[23913]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 200 was sent out Dec 8 04:44:56 /usr/sbin/kamailio[14960]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 486 was sent out Dec 5 07:46:04 /usr/sbin/kamailio[14961]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 480 was sent out Kamailio is configured to act as a proxy between SBC and a softswitch. According to the signaling we can see that these errors occure when the SBC sends CANCEL message towards Kamailio when at the same time final reply is received from the Softswitch side. The call seems to be teared down nicely and finally Kamailio replies to SBC by sending "200 ok, no more pending brances". I guess Kamailio treats this situation as an error because the sip processing engine has sent the final reply out and at the same time SBC sends the CANCEL to Kamailio. Is there any way we could handle this situation better on Kamailio end? The three example ERROR messages are from cases where B subscriber has answered the call, the B has been busy and the last one is from a case where B subscriber has been temporarily unavailable. Cheers, Olli

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is_e164 logic to detect valid number
by Patrick Wakano 11 Dec '18

11 Dec '18

Hello list, Hope you are all doing fine! I am using the is_e164() function to validate the number we receive, and I come to see that the number +555 was accepted.... After some googling it looks like(it is not very clear though) that 7 digits are the minimum we could have for e164 numbers but after checking the source code, I saw it accepts anything starting with + and having between 2 and 16 numbers. So is it really valid to have a number with just 2 digits? What is the case? Thank you, Kind regards, Patrick Wakano

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Performance tests of native vs. kemi scripting for SIP routing
by Daniel-Constantin Mierla 11 Dec '18

11 Dec '18

Hello, during the preparations of Kamailio v5.2.0 and the days after, I got the time to run some tests in order to see the differences of running similar SIP routing logic when using native configuration file scripting versus KEMI languages. So far I did it for Lua and Python (v2), hopefully I will find time to run also for the rest of supported scripting languages: Python (v3), JavaScript, Ruby, Squrrel and Lua with LuaJIT. The pleasant surprise was to see that this time, the results of using Python were similar to native scripting and Lua. When I did some basic tests during development of KEMI, using Python seemed slower. More details about how tests were performed, the config files/scripts, used tools and results were published in a wiki page: - https://www.kamailio.org/wiki/kemi/performance-tests/5.2.x A news article with further details is on kamailio website: - https://www.kamailio.org/w/2018/12/performance-tests-for-kemi-scripting-lan… Should anyhow of you have some time to run the tests in own environment and share the results (on sr-users mailing list), it would be appreciated and would help to validate the tests and discover if I forgot something in any of the configs. Note that the goal of the tests is not measuring the capacity of Kamailio in your environment, but see the differences between using different scripting languages in the same environment (so you can do the tests on raspberry pi, if you like). I would also be interested in knowing if you are using Kamailio with KEMI or you are planning to, if yes, which scripting language. If already doing it, is already in production? These are useful to plan where to allocate developer resources in the future... And to start, I am running Kamailio with Lua (4 instances) and Python (1) scripting in production deployments, with very good results so far. Cheers, Daniel -- 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 Kamailio Advanced Training -- www.asipto.com

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core at kamailio 5.1.6
by David Escartín 11 Dec '18

11 Dec '18

hello all we are seeing in a new kamailio instance some crashes. We are not sure the root cause, here you have the last core generated https://pastebin.com/xMRrug2X COuld you please take a look to see if there might be any issue with the last message that caused the core (to me seems normal) or there might be some issues related to other system libraries, etc? thanks a lot and regards david

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DTLS HA redis-pacemaker-corosync
by Жан Базаров 10 Dec '18

10 Dec '18

Hello! Are there any ways to implement using DTLS in redis HA scenario? When I reboot the active RTPEngine server the system correctly fails over to the other server and, after only a small pause (~1s), incoming audio continues to arrive. However, approximately 15 seconds after this I hear nothing but silence. Thanks.

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Kamailio World 2019 - Call for Presentations
by Daniel-Constantin Mierla 10 Dec '18

10 Dec '18

Hello, I would like to announce that Call for Presentations at Kamailio World 2019 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k07/call-for-speakers/ The 7th edition of the event takes place again in Berlin, Germany, during May 6-8, 2019. Expect over 150 participants, developers and community members as well as representatives from other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- 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 Kamailio Advanced Training -- www.asipto.com

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SIP SIMPLE TO XMPP AND VICE VERSA
by sagar malam 08 Dec '18

08 Dec '18

Hello, We have a system which using FreeSWITCH for calls and Ejabberd( XMPP) for chat.Now we want to support SIP SIMPLE as well.But we are not going to move old users to SIP SIMPLE but only new users will use SIP SIMPLE.This leads to requirement of having SIP-XMPP gateway for which are are considering Kamailio server. I have gone through documentation about Kamailio XMPP module, it is clear that Kamailio can convert SIP MESSAGE to XMPP so SIP to XMPP will work but what about XMPP to SIP MESSAGE ? Does that work as well ? Thanks in advance. -- Thanks, Sagar

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Dialplan Module limits
by Enrico Bandiera 07 Dec '18

07 Dec '18

Hi guys, does anyone know which are the limits of the diaplan module? After how many entries I will encounter serious performance issues? Thanks, Enrico.

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