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37518 discussions

Recommended openSSL version
by Igor Olhovskiy 19 Oct '22

19 Oct '22

Hello! Due to I still experience irregular Kamailio 5.4 crashes (like 1/month) related to SSL (using websockets and SIPS) I'm wondering, could openSSL upgrade change the situation? As of now in CentOS 7 I have 1.0.2k version. Does anyone have experience to fix crash-related to TLS problems with openSSL upgrade? Or maye some tuneup of TCP parameters can help here?My current setup is quite simple: children=4 enable_tls=yes tcp_accept_no_cl=yes tcp_connection_lifetime=600 tcp_max_connections=998976 # 1000000 - 1024, so we're leaving 1k for system reserve tls_max_connections=998976 Number of clients ~ 200 constantly connected to websocket. -- Best regards, Igor

5 7

IMC module
by Giovanni Iamonte 19 Oct '22

19 Oct '22

HI, I am facing a problem related to the IMC module. I have installed kamailio 5.2.x and I tried to list the members of a room-qsipchat. In my test environment the kamailio.cfg file it is very basic, the kamailio server listen on tcp=192.168.100.192:15006 the client run on a PC with IP address 192.168.100.156 the imc outbound_proxy is set modparam("imc", "db_url", DBURL) modparam("imc", "imc_cmd_start_char", "#") modparam("imc", "outbound_proxy", "sip:sip-test.quintetto.it:15006;transport=tcp") The client start by sending a #join message to kamailio and looking the log file I can see that the room-qsipchat has been create, and also I received the message 200 OK. By using the tcpdump I can see the join message from the Client to kamailio server and the reply from kamailio to the client. When I send the #list message to the room-qsipchat , I expecting to receive the message relate to room's members and the 200 OK message. When I tcp dumping I saw the following messages ******************************************************************** Request from client to list the members - from client to server ******************************************************************** TX 538 bytes Request msg MESSAGE/cseq=57652 (tdta0x77b26460a8) to TCP 192.168.100.192:15006: MESSAGE sip:room-qsipchat@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.156:48129;rport;branch=z9hG4bKPjRzgOBgY6vfKZ8Fh8GA4CWheCO840Xzud;alias Max-Forwards: 70 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it> Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57652 MESSAGE Accept: text/plain, application/im-iscomposing+xml Route: <sip:sip-test.quintetto.it:15006;transport=tcp;lr> Content-Type: text/plain Content-Length: 5 #list ******************************************************************** The server ask for the authentication - from server to client ******************************************************************** 2022-10-17 15:45:25.837 4700-10610/it.quintetto.qsipchat D/pjlib(pjsua_1): RX 574 bytes Response msg 407/MESSAGE/cseq=57652 (rdata0x77b3337330) from TCP 192.168.100.192:15006: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 192.168.100.156:48129;rport=48129;branch=z9hG4bKPjRzgOBgY6vfKZ8Fh8GA4CWheCO840Xzud;alias;received=192.168.100.156 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.e417 Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57652 MESSAGE Proxy-Authenticate: Digest realm="sip-test.quintetto.it", nonce="Y01doWNNXHVEyRtiAyN28xJdMjNIG8Ju" Server: kamailio (5.2.8 (x86_64/linux)) Content-Length: 0 ********************************************************************* The client send the authenticated request - from client to server ******************************************************************** TX 755 bytes Request msg MESSAGE/cseq=57653 (tdta0x77b26460a8) to TCP 192.168.100.192:15006: MESSAGE sip:room-qsipchat@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.156:48129;rport;branch=z9hG4bKPjj6uuy25r0kKlnICHKQWEaIweOex-sNyq;alias Max-Forwards: 70 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it> Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57653 MESSAGE Accept: text/plain, application/im-iscomposing+xml Route: <sip:sip-test.quintetto.it:15006;transport=tcp;lr> Proxy-Authorization: Digest username="quintetto00", realm="sip-test.quintetto.it", nonce="Y01doWNNXHVEyRtiAyN28xJdMjNIG8Ju", uri="sip:room-qsipchat@sip-test.quintetto.it", response="d495ec20cb1be8f1b605d4e5b1a4c357" Content-Type: text/plain Content-Length: 5 #list ************************************************************************** The server send the 200 OK message - from server to client ******************************************************************** RX 447 bytes Response msg 200/MESSAGE/cseq=57653 (rdata0x77b3337330) from TCP 192.168.100.192:15006: SIP/2.0 200 ok Via: SIP/2.0/TCP 192.168.100.156:48129;rport=48129;branch=z9hG4bKPjj6uuy25r0kKlnICHKQWEaIweOex-sNyq;alias;received=192.168.100.156 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.365f Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57653 MESSAGE Server: kamailio (5.2.8 (x86_64/linux)) Content-Length: 0 ********************************************************************************* Till here everything is OK. But regarding the members when a TCP dumping I saw a message with the room's members that go from the kamailio server to the kamailio server ********************************************************************************* MESSAGE sip:quintetto00@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.192:15006;branch=z9hG4bKf20a.be253b92000000000000000000000000.0 To: <sip:quintetto00@sip-test.quintetto.it> From: <sip:room-qsipchat@sip-test.quintetto.it>;tag=03e4c67b0f95c0aeb1f633479b4aa5cd-72f2 CSeq: 10 MESSAGE Call-ID: 37db44aa523ea24a-6567(a)192.168.100.192 Max-Forwards:.70 Content-Length: 47 User-Agent: kamailio (5.2.8 (x86_64/linux)) Content-Type: text/plain Members: *sip:quintetto00@sip-test.quintetto.it ************************************************************************************** a reply from the kamailio server to the kamailio server, with an authenticated request ************************************************************************************** SIP/2.0.407.Proxy Authentication.Required Via: SIP/2.0/TCP 192.168.100.192:15006;branch=z9hG4bKf20a.be253b92000000000000000000000000.0 To: <sip:quintetto00@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.cfc3 From: <sip:room-qsipchat@sip-test.quintetto.it>;tag=03e4c67b0f95c0aeb1f633479b4aa5cd-72f2 CSeq: 10 MESSAGE Call-ID: 37db44aa523ea24a-6567(a)192.168.100.192 Proxy-Authenticate: Digest realm="sip-test.quintetto.it", nonce="Y01fQmNNXhbsHdM1THmENyJn1MH/VD/8" Server: kamailio (5.2.8 (x86_64/linux)) Content-Length:.0 ************************************************************************************** bottom of line the outbound_proxy request for an authenticated request, but kamailio do not reply to this request. Could you please point me in the right direction? Thanks a lot for any help. Bye -- ------------------------------------------------------------------------ Ing. Giovanni Iamonte Developments and technologies area Quintetto Srl Via Monte Navale, 1 10015 - Ivrea (TO)  mobile: +39 393 9196310  tel: +39 0165 1845290  e-mail: giovanni.iamonte(a)quintetto.it  web: www.quintetto.it

2 4

ClueCon and ... FreeSWITCH 1.10.8/RHEL8 packages!
by Richard Chan 18 Oct '22

18 Oct '22

Hi sr-users, Since ClueCon is going on right now, and some Kamailio community members are involved I hope you will forgive a little promo here: I have a RHEL8 build of the just released 1.10.8 on Copr: https://copr.fedorainfracloud.org/coprs/beaveryoga/FreeSWITCH-1.10.8/ Hope you will find it useful in your kamailio + Class 5 scenarios! Cheers Shih-Ping

1 0

Testing dialog module (and acc with dialog cdrs)
by Daniel-Constantin Mierla 18 Oct '22

18 Oct '22

Hello, as I am not a user of dialog variables, I am turning to community to ask for help testing the current master branch with configurations that make use of dialog variables and acc dialog-based cdr generation. With a few reports of issues related to dialog modules and unexpected crashes, I looked over the code and noticed that the access of the value for dialog variables was not protected, making them vulnerable of invalid memory access in case of the variable was updated by another process or dialog was terminated. I introduced a couple of new functions to try to cover different use cases of getting the dlg variable values, dialog management code was not affected, but given that these commits need to be backported to stable branch (5.6), I want to get proper feedback from community that things work fine. A previous attempt of a simpler fix was not enough, having side effects to acc module for dialog-based cdr generation, because it was keeping referenced to many dlg variables at the same time. In short, it would be appreciated any feedback on testing dialog and acc with dialog-based cdr generation using git master branch. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 7-10, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

3 13

kamailio & rtpengine question
by Gerry Kernan 15 Oct '22

15 Oct '22

Hi Looking for advise if anyone could point out where im going wrong. My setup is Kamailio Ver 5.6.0 running in azure , rtpengine Version: git-master-dc395d5, behind kamailio I have asterisk ver 18.12.1 Kamailio conf I have interface set as below listen=udp:10.3.1.6:5060 advertise WANIP:5060 and rtpengine.conf set as below interface = eth0/10.3.1.5 but when I can make a call I don't get any audio, rtpengine.log extract below shows that rtp is going to the private IP rather that the Public IP of the phone that initiated the call, do I need to set rtpengine_manage() to fix this? Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Confirmed peer address as 10.3.1.5:12890 Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Kernelizing media stream: 10.3.1.5:12890 -> 10.3.1.6:30278 | 10.3.1.6:30298 -> 192.168.0.123:40042 Oct 7 09:39:00 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30279]: [core] Confirmed peer address as 10.3.1.5:12891 Kind regards gerry Gerry Kernan | Technical Director Mobile: +353(0)86-170 9790 [Graphical user interface, website Description automatically generated]<https://www.infinityit.ie/> [A picture containing text, clipart Description automatically generated,Icon Description automatically generated]

3 8

tsilo usage more generally
by Alex Balashov 14 Oct '22

14 Oct '22

Since on the topic of tsilo: 1. I watched Federico’s presentation on it from 2015, and his examples insinuate that it is necessary to store the index and label of a suspended transaction in some external vehicle, e.g. htable, for use with ts_append_to(). Why would one do this when it is possible to call ts_append() based on R-URI alone? 2. The appropriate pattern of interaction between tsilo and http_async_client — which I imagine to be a rather common use-case — is rather unclear. If I use http_async_query() to call out to a PN service, with $http_req(suspend) = 1, what should happen to the current instance of SIP request processing? That is to say: request_route { . . . t_check_trans(); if(is_method(“REGISTER”)) { . . . save(“location”, “0x06”); if(ts_append(“location”, “$tu”)) { xlog(“success”); } return; } else if(is_method(“INVITE”)) { # PN stuff. ts_store(“$var(uri)”); http_async_query(“http://url”, “PUSH_REPLY”); return; # <— ??? } } route[HTTP_PUSH_REPLY] { . . . if(!t_relay()) sl_reply_error(); } I should add that this proxy front-ends a registrar, but is not in itself a registrar. It contains a shadow registrar - hence the save() calls - in order to be able to take advantage of the ’tsilo’ module, given its dependency on registrar/usrloc. This leads to the difference that the gateway receives requests with RURIs that have already been resolved to a contact binding elsewhere, so there are no lookup() calls. How should this play into the http_async_client workflow? Appreciate any help! — Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

tsilo logging
by Alex Balashov 13 Oct '22

13 Oct '22

Hi, When there are no stored transactions for an AOR, it logs this error: Oct 13 07:48:18 gw1 /usr/sbin/kamailio[14758]: ERROR: tsilo [ts_append.c:64]: ts_append(): failed to retrieve record for sip:abalashov@sip.evaristesys.com But not having transactions ts_store()’d is a rather normal occurrence, right? It is the expected case the vast majority of the time that someone registers. Does that mean one is simply destined to have this scary-looking error, to the syslog ERROR facility, for a normal runtime condition? Is there not some principle against using attention-grabbing error messages this way? Would a patch to fix be welcome? — Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

About kamailio + freeswitch issue on Kamailio v5.6
by SparkleZou 12 Oct '22

12 Oct '22

Hi Sir/Mada, I'm trying to integrate kamailio + freeswitch, according to the manual http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc on Kamailio v5.6 modified the kamailio.cfg file, add WITH_FREESWITCH parts, then kamailio could NOT start up. Could you please help check, what should be modified to fit the version 5.6？ Thanks！ BR， Sparkle Zou

4 9

Re: [SR-Users] play_media sounds interrupted
by Stepan Kislyakov 12 Oct '22

12 Oct '22

> The duration of the media playback is returned from the play_media() > command (see `media_duration` modparam). You could use some sort of > timer to invoke media unblocking after media playback ends. Sadly my > Kamailio-Fu isn't strong enough to be able to tell you how to run this > sort of timer. :) > > Cheers Thank you for the hint. I was thinking about timers, but rtimer cannot be triggered in an event route. I used a "timer" module, it could trigger another route from the event route but when a new route is triggered unblock_media() is not aware of the call-id and if i assign call-id to a variable in the event route, that variable is unavailable at new route. Hope that make sense. here is my peace of code: loadmodule "timer.so" modparam("timer", "declare_timer", "PB_TIMER=START,5000"); route[START]{ xlog("current call id at event route is $ci, but call-id avp is $avp(call_id)"); unblock_media("all"); timer_enable("PB_TIMER",0); #we need to disable timer or it will be triggered over and over again } event_route[dialog:start]{ xlog("L_INFO", "dialog started, received 200OK"); #block_media(); play_media("all file=/usr/local/src/myfile.wav"); $avp(call_id)=$ci; timer_enable("PB_TIMER",1); }

1 0

play_media sounds interrupted
by Stepan Kislyakov 11 Oct '22

11 Oct '22

Hi, I'm using latest kamailio 5.6.2 and rtpengine. I need to play announcement to both parties when call is answered by callee. Now I run play_media in event route but when audio start announcement sounds like we have a 50% packet loss. After it is completed there is no problem with audio both sides. I figured out that if i do block_media() my announcement play perfectly but i dont understand how to do unblock_audio() after announcement ends. Appriciate any help! here is my code: event_route[dialog:start]{ xlog("L_INFO", "dialog started, received 200OK"); block_media(); play_media("all file=/usr/local/src/myfile.mp3"); #unblock_media(); } -- Best Regards Stepan Kislovich

2 1

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