sr-users September 2025

sr-users@lists.kamailio.org

42 participants
40 discussions

Re: using dlg_refer
by Robert Boardman 27 Sep '25

27 Sep '25

Hi, Thanks for responding 1. Debug logs attached 2. I'm not seeing any attempt to refer in the pcap traces , or sngrep I think it's just quietly not sending the dlg_refer for some reason?? Rob Robert Boardman Solution Architect Tel: 03330147908 Email: Robert.Boardman(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at The Scalpel, 18th Floor, 52 Lime Street, London EC3M 7AF and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

2 1

tm stats
by Alexis Fidalgo 26 Sep '25

26 Sep '25

Hello all, im getting tm.stats thru rpc { "jsonrpc": "2.0", "result": { "current": 594, "waiting": 585, "total": 3135603, "total_local": 0, "rpl_received": 0, "rpl_generated": 6271197, "rpl_sent": 6271197, "6xx": 0, "5xx": 1, "4xx": 0, "3xx": 3135593, "2xx": 0, "created": 3135603, "freed": 3135009, "delayed_free": 0 }, "id": 1121 } is there a way to include this stats in xhttp_prom? Thanks, regards

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http_async_client
by David Villasmil 26 Sep '25

26 Sep '25

Hello guys, I'm trying to use http_async_client to send a json to a server and it's supposed to use $http_req(body) to set the content... there's no way i can make it work: /usr/sbin/kamailio[3197588]: INFO: <script>: HTTP body: {"from":"XXXX","to":["XXXX"],"call_id":"fJYJFHAgao","timestamp":"2025-09-25T20:14:52Z"} <- my var /usr/sbin/kamailio[3197588]: INFO: <script>: HTTP body: <null> <- $http_req cfg: xlog("L_INFO", "HTTP body: $var(body)\n"); $http_req(body) = $var(body); xlog("L_INFO", "HTTP body: $http_req(body)\n"); It seems to me that $http_req(body) is write only, BUT on the web server: {"error":"Invalid request: missing required fields (from, to)","received":{}} [353] Help is appreciated, thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com

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Receiving WS keepalives
by Alex Balashov 25 Sep '25

25 Sep '25

As far as I understand it, the keepalive processes in the `websocket` module concern processes dedicated to scheduling outgoing CRLF pings to clients. What if the clients are initiating the ping, and the WS module responds with a pong? I assume the keepalive process pool does not handle that, but which processes do? Is it the core timer process, or the TLS receiver? Suspected issue: insufficient resources to respond to an enormous amount of \n pings, which are sent every 2 seconds by every client. Which resources would one increase to help with this? Cheers, -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com, https://www.csrpswitch.com Tel: +1-706-510-6800

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Unix Sockets in Go
by mohsen khashei 24 Sep '25

24 Sep '25

Hi I want to run rpc commands like permissions.addressReload from a go application but with no luck please help me my code is: cmd:="permissions.addressReload" conn, err := net.Dial("unix", "/run/kamailio/kamailio_ctl") if err != nil { return "", fmt.Errorf("connect error: %w", err) } defer conn.Close() // _, err = fmt.Fprintf(conn, "%s\n", cmd) _, err = conn.Write([]byte(cmd)) if err != nil { return "", fmt.Errorf("write error: %w", err) } scanner := bufio.NewScanner(conn) var output string for scanner.Scan() { output += scanner.Text() + "\n" } if err := scanner.Err(); err != nil { return "", fmt.Errorf("read error: %w", err) } ` the only thing i get is: ¡0#^@ ^A<90><91>^^bad request: corrupted packet^@

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db_redis has some problem
by yukino5506＠163.com 24 Sep '25

24 Sep '25

hello everyone,i want to use the acc module to record conversation history and write it to redis. Now I have encountered a problem when integrating redis in kamailio. Even after I downloaded the hiredis library from github and recompiled and installed kamailio, I still cannot solve the problem. The log is as follows: 0(47879) ERROR: <core> [core/sr_module.c:613]: ksr_load_module(): could not open module </usr/local/kamailio/lib64/kamailio/modules/db_redis.so>: /usr/local/kamailio/lib64/kamailio/modules/db_redis.so: undefined symbol: redis_supports_expires 0(47879) CRITICAL: <core> [core/cfg.y:4086]: yyerror_at(): parse error in config file /usr/local/kamailio/etc/kamailio/kamailio.cfg, line 282, column 12-24: failed to load module Please let me know if any necessary information is missing Thank you in advance

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Webrtc->SIP Audio Issues
by Fernando Lopes 22 Sep '25

22 Sep '25

Hello everyone, I'm currently trying to make WebRTC-to-SIP calls. My setup is: WebRTC → WSS → Kamailio → UDP → Asterisk, using rtpengine. I have a route named GET_ICE that is called in onreply_route. Everything else works: SIP-to-SIP calls and WebRTC messaging are fine. Only WebRTC-to-SIP calls have audio issues. I've tested many different ICE flags, but without success. In my testing I think I've managed to get audio to kamailio but not to webrtc, but decided to rollback to start to get help. In my PJSIP configuration, I allow these codecs: allow=g722,ulaw,vp9,vp8,h264 My Kamailio code: route[GET_ICE] { if ( $proto == "wss" ) { if (is_ip_rfc1918("$si")) { $dlg_var(ice) = " "; } else { $dlg_var(ice) = " ICE=remove RTP "; } } else { $dlg_var(ice) = "ICE=FORCE"; } } route[BUILD_RTPENGINE_PARAMETER] { route(SET_RTP_DIRECTION); # returns $dlg_var(rtp_direction) if ( is_method("INVITE") ) { route(GET_ICE); # returns $dlg_var(ice) } xlog("L_WARN", "[BUILD_RTPENGINE_PARAMETER] > $dlg_var(rtp_direction)\n"); $var(rtp_param) = "replace-origin replace-session-connection " + $dlg_var(ice) + " " + $dlg_var(rtp_direction); xlog("L_WARN", "[RTPENGINE]: Method $rm status $rs protocol: $dlg_var(call_type) -> $dlg_var(rtp_direction) -> [$dlg_var(ice)]\n"); xlog("L_WARN", "[RTPENGINE]: Param: [$var(rtp_param)]\n"); rtpengine_manage($var(rtp_param)); } My logs: 3(47) WARNING: <script>: [REPLY] 1xx or 2xx received 3(47) WARNING: <script>: [SET_RTP_DIRECTION] -> reply from internal = direction=internal direction=external 3(47) WARNING: <script>: [BUILD_RTPENGINE_PARAMETER] > direction=internal direction=external 3(47) WARNING: <script>: [RTPENGINE]: Method INVITE status 200 protocol: <null> -> direction=internal direction=external -> [] 3(47) WARNING: <script>: [RTPENGINE]: Param: [replace-origin replace-session-connection direction=internal direction=external] [1758217893.108564] INFO: [ID="m7e648isjm9n96ti62k9"]: [control] Received command 'answer' from 10.5.0.8:42938 [1758217893.108583] INFO: [ID="m7e648isjm9n96ti62k9"]: [core] replace-session-connection flag encountered, but not supported anymore. [1758217893.108998] WARNING: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1"]: [core] ICE restart detected, but reset not allowed at this point [1758217893.109379] INFO: [ID="m7e648isjm9n96ti62k9"]: [control] Replying to 'answer' from 10.5.0.8:42938 (elapsed time 0.000804 sec) [1758217893.109411] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [ice] ICE negotiated: peer for component 1 is 10.5.0.2:28384 [1758217893.109418] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [ice] ICE negotiated: local interface 10.5.0.8 [1758217893.208990] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [ice] ICE negotiated: peer for component 1 is 10.5.0.2:28384 [1758217893.209036] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [ice] ICE negotiated: local interface 10.5.0.8 [1758217894.111112] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [crypto] DTLS: Peer certificate accepted [1758217894.112292] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 [1758217894.112341] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 [1758217896.299109] INFO: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [core] Confirmed peer address as 10.5.0.2:28384 [1758217896.299174] ERR: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [srtp] SRTP output wanted, but no crypto suite was negotiated [1758217901.298677] ERR: [ID="m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828" index="1" port="14656"]: [rtcp] SRTCP output wanted, but no crypto suite was negotiated [1758217930.743494] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] Closing call due to timeout [1758217930.743636] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] Final packet stats: [1758217930.743651] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] --- Tag '2bad2cacef0945b1a506a198aeb4ae01', created 1:00 ago for branch '' [1758217930.743659] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec [1758217930.743670] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] --------- Port 10.5.0.8:15028 <> 172.31.208.1:4000 , SSRC 0, in 0 p, 0 b, 0 e, 60 ts, out 0 p, 0 b, 0 e [1758217930.743677] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] --------- Port 10.5.0.8:15029 <> 172.31.208.1:4001 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 60 ts, out 0 p, 0 b, 0 e [1758217930.743683] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] --- Tag '88cf8c49-cdfc-475e-8075-a3ce0a36ecca', created 1:00 ago for branch '' [1758217930.743687] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec [1758217930.743693] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] --------- Port 10.5.0.8:17956 <> 10.5.0.2:64570, SSRC 0, in 0 p, 0 b, 0 e, 60 ts, out 0 p, 0 b, 0 e [1758217930.743699] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] --------- Port 10.5.0.8:17957 <> 10.5.0.2:64571 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 60 ts, out 0 p, 0 b, 0 e Thank you.

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Proposal for a more granular configuration of DMQ
by Björn Klasen 19 Sep '25

19 Sep '25

Hello I'm testing a lot with DMQ and its possibilities for replicating data between several nodes. We also use DMQ in production for replicating usrloc, htable and dialog. So far so good. But in my log I regularly see the following message: ERROR: dialog [dlg_dmq.c:293]: dlg_dmq_handle_msg(): dialog [22051:221142] not found As I tried to troubleshoot this message I discovered the following: On network everything is OK. The KDMQ messages are sent and received in the correct order. But they are not processed by the DMQ system in Kamailio in the correct order. It might happen that the dialog KDMQ message for the initial INVITE is not processed before the KDMQ message for the provisional response. I repeated the tests in my lab with different number of workers for DMQ. But after I set the number of workers to 1 everything seems to be fine. But now comes the problem. As I mentioned, we also replicate htable between the nodes, so we can use all nodes as hot standby. If I raise the CPS one worker is not enough to handle htable and dialog and userloc. So I thought it might be cool to have the possibility to set the amount of worker for the modules separately. So I would just use one worker for dialog and all other workers for any other modules that use DMQ because in my opinion dialog is the only thing where the order of the KDQM messages matters. As I'm not capable of programming things like that I wanted just put this a think for further developments of DMQ. I'm really looking forward to your thoughts about this. With best regards from northern Germany Björn -- Björn Klasen, Teamleitung NGN VoIP-Backbone TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de<mailto:bklasen@tng.de> https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO), Sabine Domes, Gunnar Peter, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 13 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

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Dispatcher module - pre-set route header (send via proxy?)
by Benoit Panizzon 18 Sep '25

18 Sep '25

Hi all Kamailio <=> SBC <=> Interconnection 10.1.1.1 <=> i: 10.1.1.2 SBC e: 10.99.1.77 <=> IC: 10.88.88.1 |> i: 10.1.1.3 SBC e: 10.99.1.77 <=> IC: 10.88.88.2 At the moment I have the issue that our commercial SBC can only handle n to 1 static ingress to egress mappings. So basically I can tell the SBC: Do route everything you get from Kamailio 10.1.1.1 received in 10.1.1.2 to IC peer: 10.88.88.1 and vice versa. So I need one internal SBC IP per ic peer. But we want redundancy and load balancing. So IC peer (like the largest telco in Switzerland) increased their redundancy and are reachable on 4 ip addresses via two independent exchanges. Let's use 10.88.88.1 to 10.88.88.4 I discovered that our SBC would support Route header. At the moment my dispatcher.lst looks like this: 1100 sip:10.1.1.2:5060 0 0 name="IC-A" 1100 sip:10.1.1.3:5060 0 0 name="IC-A" But! Is there a possibility to set a route on the dispatcher list? Like: 1100 sip:10.88.88.1:5060 Route:10.1.1.2:5060 0 0 name="IC-A" 1100 sip:10.88.88.2:5060 Route:10.1.1.2:5060 0 0 name="IC-A" 1100 sip:10.88.88.3:5060 Route:10.1.1.2:5060 0 0 name="IC-A" 1100 sip:10.88.88.4:5060 Route:10.1.1.2:5060 0 0 name="IC-A" So I could use only one ip address on the internal SBC interface. And I could extend this to further interconnections using only one IP and have the SBC route according the route header received: 1100 sip:10.88.88.1:5060 Route:10.1.1.2:5060 0 0 name="IC-A" 1100 sip:10.88.88.2:5060 Route:10.1.1.2:5060 0 0 name="IC-A" 1100 sip:10.88.88.3:5060 Route:10.1.1.2:5060 0 0 name="IC-A" 1100 sip:10.88.88.4:5060 Route:10.1.1.2:5060 0 0 name="IC-A" 1200 sip:10.5.5.10:5060 Route:10.1.1.2:5060 0 0 name="IC-B" 1200 sip:10.5.5.10:5060 Route:10.1.1.2:5060 0 0 name="IC-B" 2000 sip:10.1.1.10:5060 0 0 name="Kamailio-Registrar-A" 2001 sip:10.1.1.11:5060 0 0 name="Kamailio-Registrar-B" 3000 sip:10.1.1.20:5060 0 0 name="Voicemail-A" 3000 sip:10.1.1.21:5060 0 0 name="Voicemail-B" and so on to send call to the various IC via the same SBC ip with pre-set route? The obproxy attribute is only for pings, right? Mit freundlichen Grüssen -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

Sugestions on Kamailio learning Courses
by Fernando Lopes 18 Sep '25

18 Sep '25

Hello everyone, I’m interested in starting a Kamailio course, but I’m not sure where to find one. The "only" course I’ve come across so far is on Udemy. Does anyone have other suggestions or recommendations? I'm currently using the video sessions on youtube of kamailio world to learn some things and this forum. Thank you!

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sr-users September 2025