sr-users September 2025

sr-users@lists.kamailio.org

42 participants
40 discussions

Active-Active notifier using presence module
by i.ribakov＠zaleos.net 17 Sep '25

17 Sep '25

I would like to configure 2 (or more) Kamailio to act as HA notifier cluster. I want subscriptions to survive complete node reboots so I'm looking at using presence module with DB replication, specifically: subs_db_mode = 3 and publ_cache = 0. However while running some initial tests I came across the table locking issue (MySQL error 1100). I'm using MariaDB 10.11 with innoDB and default root user. Further research led me to an older thread (https://lists.kamailio.org/pipermail/sr-users/2019-January/104288.html) and the documentation for db_table_lock_type parameter that states that it should be set to 0 (no locking) when using DB-only mode (subs_db_mode = 3). My question is - if presence module is not using table locks, what prevents notify processes on 2 or more different nodes to simultaneously process the same presentity record and issue multiple notifies for the same subscription?

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how to kamcmd ul.add ?
by Benoit Panizzon 17 Sep '25

17 Sep '25

Hi Gang Recently we got a request from a customer, if we could just route calls to his number to an arbitrary sip URI. First I was thinking on the lines - yes I could probably append_branch() yet another contact from yet another database table. But then I came across static entries in the userloc database, that would be even easier! Just add the additional entry with kamcmd and done. https://www.kamailio.org/docs/modules/5.8.x/modules/usrloc.html#usrloc.r.add Only issue - I haven't figured out how to add that contact - I would also need to add a path to send the invite via our rtpengine SBC. # kamcmd ul.add location AOR-XY sip:customer@example.com 0 \ 440 sip:sbcip:5060 0 0 0 0 0 I am aware, the path should contain ;lr But no matter what I try I get: error: 500 - Not enough parameters or wrong format Does anyone have an example on how to add a static contact to the location database? I managed to add a contact with kamctl ul add but I don't manage to add the required path or q value. 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 ______________________________________________________

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siptrace: sip_trace_mode("d") not working
by mm e 17 Sep '25

17 Sep '25

version: V6.0.3 loadmodule "siptrace.so" modparam("siptrace", "trace_to_database", 1) modparam("siptrace", "db_url", DBURL) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_dialog_ack", 1) modparam("siptrace", "trace_dialog_spiral", 1) #modparam("siptrace", "trace_mode", 2) modparam("siptrace", "trace_db_mode", 1) if (is_method("INVITE")) { dlg_manage(); setflag(22); sip_trace_mode("d"); route(RELAY); } ERROR: trace_add_info_xavp(): URI should be in raw format here sip_trace_helper(): failed to serialize siptrace info! Won't trace dialog!

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Wrong Contact Header
by Fernando Lopes 16 Sep '25

16 Sep '25

Hello everyone, I’m running into an issue with Kamailio (exposed to the public) talking to Asterisk. The problem only happens with UDP/TCP SIP (not with WebRTC), and it’s related to the 200 OK response from Asterisk. Asterisk is sending the 200 OK with its internal IP address in the Contact header, which Kamailio forwards to the client. As a result, the client tries to send the ACK directly to the internal IP, and it fails. Logs (WebRTC → Kamailio → Asterisk) || In sngrep, the connection shows as: 10.5.0.8:5060 (Kamailio) <-> 10.5.0.2:5060 (Asterisk) SIP/2.0 200 OK Via: SIP/2.0/UDP 172.31.217.74:5060;rport=5060;received=10.5.0.8;branch=z9hG4bK0186.15eb7eeb80260f12a3a4d45264371b16.0 Via: SIP/2.0/WSS 05gol16panaa.invalid;rport=65443;received=172.31.208.1;branch=z9hG4bK2464299 Call-ID: i0c08fra5hdplpdj4pq4 From: "916c67e8-652f-49ea-b288-2bbbee8dbf21" <sip:User1@172.31.217.74>;tag=86ca2rhiqs To: <sip:7@172.31.217.74>;tag=b012b645-bf96-4fba-9166-f7a00b10290b CSeq: 2 INVITE Server: Asterisk PBX 18.24.3 Contact: <sip:10.5.0.2:5060> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 667 ACK sip:10.5.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 172.31.217.74:5060;branch=z9hG4bK0186.94970c4407431a3133bec46208b4be06.0 Via: SIP/2.0/WSS 05gol16panaa.invalid;rport=65443;received=172.31.208.1;branch=z9hG4bK3452625 To: <sip:7@172.31.217.74>;tag=b012b645-bf96-4fba-9166-f7a00b10290b From: "916c67e8-652f-49ea-b288-2bbbee8dbf21" <sip:User1@172.31.217.74>;tag=86ca2rhiqs CSeq: 2 ACK Call-ID: i0c08fra5hdplpdj4pq4 Max-Forwards: 69 Supported: outbound User-Agent: SIP.js/0.21.1 Content-Length: 0 Logs (TCP call using MicroSIP → Kamailio → Asterisk) SIP/2.0 200 OK Via: SIP/2.0/UDP 172.31.217.74:5060;rport=5060;received=10.5.0.8;branch=z9hG4bK82df.7f15d0606c2de9f083b2757842de918f.0;i=1 Via: SIP/2.0/TCP 172.31.208.1:57670;rport=57670;received=172.31.208.1;branch=z9hG4bKPj379b3f941943461182af8d639cd93122;alias Call-ID: 5fbb1276c10e4041b9d5bc746bd48035 From: "6001" <sip:6001@localhost>;tag=009616b957db491cbea95fe3bfc123ee To: <sip:100@localhost>;tag=7345097c-95c0-42da-9b8e-1f39c5ec7f53 CSeq: 4449 INVITE Server: Asterisk PBX 18.24.3 Contact: <sip:10.5.0.2:5060> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 481 Any suggestions on how to fix this problem. I'm using a modified version of this: https://github.com/davidcsi/kamailio-private-public/blob/master/kamailio.cfg Thank you.

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Best SVC methods/ways for security
by Fernando Lopes 15 Sep '25

15 Sep '25

Hello everyone, I would like to improve the security of my setup. At the moment, I only have pike enabled for blocking spam, but I’d like to extend this with additional measures. In particular, I’m interested in implementing features such as: IP/domain whitelisting and blacklisting Any other recommended security modules or best practices for Kamailio Has anyone here already set up a similar security layer? What would you recommend as the most effective approach? Thank you.

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send_reply not accepting values
by Sergio Charrua 11 Sep '25

11 Sep '25

Hi all! I'm using EVAPI module. Upon INVITE, Kamailio gets a list of routes from a script using *evapi_async_relay*() . The EVAPI script (Python) then returns a RTJson structure. The Kamailio logic is set to iterate through each route and if at the end all routes have failed (ex.: if all egress endpoints are unavailable), then sends a EVAPI request using *evapi_relay*() to the script. Note that this request is not asynchronous (using evapi_async_relay() would fail with error stating that there is no transaction available....). This is my evapi:message-received: event_route[evapi:message-received] { # Parse the message format: RESPONSE:index:label:data $var(msg) = $evapi(msg); #$var(logmsg) = $(var(msg){s.substr,0,100}); xlog("L_INFO", "evapi:message-received - Received EVAPI message: $var(msg) ...\n"); $var(type) = $(var(msg){s.select,0,:}); $var(tindex) = $(var(msg){s.select,1,:}{s.int}); $var(tlabel) = $(var(msg){s.select,2,:}{s.int}); $var(response) = $(var(msg){re.subst,/^[^:]*:[^:]*:[^:]*://}); xlog("L_INFO", "Part 0: '$var(type)'\n"); xlog("L_INFO", "Part 1: '$var(tindex)'\n"); xlog("L_INFO", "Part 2: '$var(tlabel)'\n"); xlog("L_INFO", "Part 3: '$var(response)'\n"); if ( $var(type) == "RESPONSE" && $var(tindex) =~ "^[0-9]+$" && $var(tlabel) =~ "^[0-9]+$" ) { xlog("L_INFO", "evapi:message-received - RESPONSE index $var(tindex) label $var(tlabel) \n"); xlog("L_INFO", "evapi:message-received - Continue Call processing\n"); if (jansson_get("action", $var(response), "$var(action)")){ if ($var(action) == "CALL_REJECT"){ route(EVAPIRESPONSE); } } # Resume transaction t_continue("$var(tindex)", "$var(tlabel)", "EVAPIRESPONSE"); } else { xlog("L_INFO", "evapi:message-received - invalid message received from EVAPI. Cancelling the call \n"); t_reply("500", "Internal Server Error"); exit; } } CALL_REJECT action is received after the *evapi_relay*() request to the EVAPI script (python) that is launched after a route_failure. For example, Kamailio requests for error mappings when a call has been tried and no egress endpoint was reachable. My EVAPIRESPONSE route block is: route[EVAPIRESPONSE] { xlog("L_INFO", "EVAPIRESPONSE - Resuming INVITE processing with index $var(tindex) label $var(tlabel)\n"); xlog("L_INFO", "EVAPIRESPONSE - JSON : $var(response) \n"); # Test each condition individually if ($var(response) == "FALSE" or $var(response) == "NO_ROUTES") { # Check if response is FALSE (drop the call) xlog("L_ERR", "EVAPIRESPONSE - Python script returned FALSE or NO ROUTES, dropping call\n"); $var(code) = 500; $var(text) = "Internal Server Error"; route(REPLY_SAFE); exit; } if (jansson_get("action", $var(response), "$var(action)")){ if ($var(action) == "DROP") { jansson_get("error_code", $var(response), "$var(error_code)"); jansson_get("error_text", $var(response), "$var(error_text)"); xlog("L_INFO", "EVAPIRESPONSE - dropping call with returned error $var(error_code) - $var(error_text) \n"); $var(code) = $var(error_code); $var(text) = $var(error_text); route(REPLY_SAFE); }else if ($var(action) == "CALL_REJECT") { jansson_get("reject", $var(response), "$var(sip_error)"); if ( "$var(sip_error)" == "" ) { xlog("L_INFO", "EVAPIRESPONSE - CALL_REJECT with no specific error code; return the received error\n"); return; } else { jansson_get("error_code", $var(sip_error), "$var(error_code)"); jansson_get("error_text", $var(sip_error), "$var(error_text)"); xlog("L_INFO", "EVAPIRESPONSE - CALL_REJECT with $var(error_code) - $var(error_text)\n"); $var(code) = $var(error_code); $var(text) = $var(error_text); route(REPLY_SAFE); } } } # Process RTJSON routing if (jansson_get("rtjson", $var(response), "$var(rtjson_str)")) { jansson_get("retry_on_failure", $var(response), "$dlg_var(retry_on_failure)"); xlog("L_DEBUG", "EVAPIRESPONSE - retry_on_failure = $dlg_var(retry_on_failure) \n"); $dlg_var(rtjson_str) = $var(rtjson_str); # Process routes from RTJSON route(RTJSON_PROCESS); } else { xlog("L_ERR", "EVAPIRESPONSE - Invalid response format from Python\n"); $var(code) = 500; $var(text) = "Internal Server Error"; route(REPLY_SAFE); exit; } } # Tiny helper to reply correctly inside/outside TM route[REPLY_SAFE] { # validate code (default to 480 if bad/empty) if (!($var(code) =~ "^[1-6][0-9][0-9]$")) $var(code) = 480; if ($var(text) == "") $var(text) = "Temporarily Unavailable"; # remove CR/LF that can break the reason phrase $var(text) = $(var(text){re.subst,/[\r\n]/ /g}); if (t_check_trans()) { xlog("L_INFO", "REPLY_SAFE - replying with t_reply\n"); t_reply("$var(code)", "$var(text)"); } else { xlog("L_INFO", "REPLY_SAFE - replying with send_reply\n"); send_reply("$var(code)", "$var(text)"); } exit; } I have tried with t_reply, sl_send_reply, send_reply but the SIP trace only shows SIP 503 Service Unavailable being sent from Kamailio to the ingress endpoint. Setting log to DEBUG level doesn't show anything neither: Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169257 457395 1 1 OPTIONS 123} <script>: evapi:message-received - Received EVAPI message: RESPONSE:2825:1816929401:{"action": "CALL_REJECT", "call_id": "338632827805594c569838281a742c0b@10.20.0.5:5060", "reject": {"error_code": 480, "error_text": "some text"}} ... Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169485 457395 1 1 OPTIONS 123} <script>: Part 0: 'RESPONSE' Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169497 457395 1 1 OPTIONS 123} <script>: Part 1: '2825' Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169506 457395 1 1 OPTIONS 123} <script>: Part 2: '1816929401' Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169514 457395 1 1 OPTIONS 123} <script>: Part 3: '{"action": "CALL_REJECT", "call_id": "338632827805594c569838281a742c0b@10.20.0.5:5060", "reject": {"error_code": 480, "error_text": "some text"}}' Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169639 457395 1 1 OPTIONS 123} <script>: evapi:message-received - RESPONSE index 2825 label 1816929401 Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169654 457395 1 1 OPTIONS 123} <script>: evapi:message-received - Continue Call processing Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169816 457395 1 1 OPTIONS 123} <script>: EVAPIRESPONSE - Resuming INVITE processing with index 2825 label 1816929401 Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169829 457395 1 1 OPTIONS 123} <script>: EVAPIRESPONSE - JSON : {"action": "CALL_REJECT", "call_id": " 338632827805594c569838281a742c0b@10.20.0.5:5060", "reject": {"error_code": 480, "error_text": "some text"}} Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.169998 457395 1 1 OPTIONS 123} <script>: EVAPIRESPONSE - CALL_REJECT with 480 - some text Sep 5 19:16:48 ire-lab-kamailio1 kamailio[457395]: INFO: {1757092608.170253 457395 1 1 OPTIONS 123} <script>: REPLY_SAFE - replying with send_reply And nothing helps is found in logs. The goal is to send a custom SIP error and message depending if the error_code and error_text is specified. If not, then the message received from egress endpoint should be forwarded to the ingress endpoint. I just can't understand why the response is always SIP 503. Here is a SIP trace: [image: image.png] Any suggestions? Thanks in advance. Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

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how to use is_socket_name()
by Denys Pozniak 11 Sep '25

11 Sep '25

Hey! I don't understand how the *is_socket_name()* function works. I assigned a name to each socket, but the function works regardless of which socket the traffic arrives at. *# kamailio -c* *Listening on* * udp: 1.1.1.1 [1.1.1.1]:5060 name sip_fl_udp* * tcp: 1.1.1.1 [1.1.1.1]:5060 name sip_fl_tcp* *script:* * if ( is_socket_name("sip_fl_udp") ) {* * xlog("L_WARN", "[test] * proto=$proto -> sip_fl_udp \n");* * } else if ( is_socket_name("sip_fl_tcp") ) {* * xlog("L_WARN", "[test] * proto=$proto -> sip_fl_tcp \n");* * }* *syslog:* *<script>: [test] * proto=udp -> sip_fl_udp* *<script>: [test] * proto=tcp -> sip_fl_udp* -- BR, Denys Pozniak

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Kamailio v6.0.3 Released
by Daniel-Constantin Mierla 10 Sep '25

10 Sep '25

Hello, Kamailio SIP Server v6.0.3 stable release is out! This is a maintenance release of the latest stable branch, 6.0, that includes fixes since the release of v6.0.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v6.0.x. Deployments running previous v6.0.x versions are strongly recommended to be upgraded to v6.0.3. For more details about version 6.0.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/09/kamailio-v6-0-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v6.0.3
by Daniel-Constantin Mierla 10 Sep '25

10 Sep '25

Hello, I am considering to release Kamailio v6.0.3 (out of branch 6.0) on Wednesday, Sep 10, 2025. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

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SIP Diversion Problem
by ale Gural 10 Sep '25

10 Sep '25

Hello, I am using version: kamailio 5.8.6 I try to send in INVITE message the follow Diversion header, Diversion: "39988246 39988246" <sip:123@10.25.148.41 >;privacy=off;reason=unconditional;counter=1,<sip:11111111@10.25.148.41>,< sip:39946001@10.25.148.41 ;user=phone>;privacy=off;reason=unconditional;counter=1 in my script code I need to separate every diversion Multibody wit the follow (example): $var(i) = 0; while($var(i) < $hflc(Diversion) ) { remove_hf("Diversion"); $vn(new_div) = $(hfl(Diversion)[1]); $var(i) = $var(i) +1; } when $var(i) = 1, the kamailio make restart with exit by signal 11 I appreciate your help!

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sr-users September 2025