- sr-users - kamailio.org

Releasing v5.7.7
by Daniel-Constantin Mierla 21 May '25

21 May '25

Hello, I am considering to release Kamailio v5.7.7 (out of branch 5.7) on Wednesday, May 21, 2025, as the last one from the 5.7.x series, to mark the end of support for it. Currently 5.8.x and 6.0.x are the latest two maintained branches. 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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Re: stir/shaken integration with clearip redirect method
by JR Richardson 20 May '25

20 May '25

I appreciate all the feedback Fellas, checking the docs, there is no need to change the RURI, they do what some E164 fixup presented without the +. I'll follow up after testing, got a bit to chew one here. Thanks! JR Richardson

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stir/shaken integration with clearip redirect method
by JR Richardson 19 May '25

19 May '25

Hey Folks, I'm integrating STIR/SHAKEN using kamailio, but not using self hosted certs with the secsipid module, just using a redirect from clearip on the SIP Calls. I've got the basic connection to clearip using t_relay_to_tcp for testing, but I'm struggling with the checks and timeouts. I have call progress past the initial request where I need to wait for a response. Should I put the clearip invite in its own route block use return? Has anyone successfully integrated the redirect method with kamailio to clearip and can give me some example route checks in the kamailio configs? Thanks. JR

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Server closing TLS connection
by Ihor Olkhovskyi 16 May '25

16 May '25

Hello, I'm having a strange issue, when Kamailio(?) is closing TLS connection just in the middle of session. Usually I see a sort of timeout ~ 600 secs. But it does not ring a bell for me. With strace I've found, that it's actually not Kamailio, but underlying SSL library is doing this, according to line I've got from strace [2025-05-16 10:00:30] PID=1337952 FD=0 SYSCALL=close -> [No socket info] Raw: sendto(3, "<183>May 16 10:00:30 /usr/local/sbin/kamailio[1337952]: DEBUG: tls [tls_server.c:712]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7f01a1ab1870\n", 150, MSG_NOSIGNAL, NULL, 0) = 150 As I don't see any explicit calls of tls_h_tcpconn_close_f in the source code, I assume it's a callback from the external OpenSSL(?) module. What would be the best way to debug why OpenSSL(?) is terminating connection, cause on Wireshark I just do see usual TCP connection with keepalives [image: image.png] kamailio -v version: kamailio 5.8.3 (x86_64/linux) 6f8a04 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_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. openssl --version OpenSSL 3.2.2 4 Jun 2024 (Library: OpenSSL 3.2.2 4 Jun 2024) enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=30 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=5 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_script_mode=1 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 Many thanks in advance! -- Best regards, Ihor (Igor)

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extract_candidate() errors on 6.0.0
by Arsenijs Kabihno 14 May '25

14 May '25

Hello, I am getting the following errors on kamailio 6.0.0: May 14 07:10:20 hostname.xxxxxx.net docker[20678]: 53(59) ERROR: {2 24186 INVITE 7c127e81-45a8-41bb-ab64-7b0e7223f7ff} <core> [core/parser/sdp/sdp_helpr_funcs.c:336]: extract_candidate(): no component in `a=candidate' May 14 07:10:20 hostname.xxxxxx.net docker[20678]: 53(59) ERROR: {2 24186 INVITE 7c127e81-45a8-41bb-ab64-7b0e7223f7ff} <core> [core/parser/sdp/sdp_helpr_funcs.c:336]: extract_candidate(): no component in `a=candidate' May 14 07:10:20 ap07stvp.dyninno.net docker[20678]: 53(59) ERROR: {2 24186 INVITE 7c127e81-45a8-41bb-ab64-7b0e7223f7ff} <core> [core/parser/sdp/sdp_helpr_funcs.c:336]: extract_candidate(): no component in `a=candidate' Candidates in SDP looks valid per RFC: a=candidate:vnyxKH3f79Lw0zSj 1 UDP 2130706431 1x.xxx.xxx.xxx 26010 typ host a=candidate:vnyxKH3f79Lw0zSj 2 UDP 2130706430 1x.xxx.xxx.xxx 26011 typ host a=remote-candidates:1 9x.xx.xxx.xxx 56628 I am using the 6.0.0-alpine version (https://github.com/kamailio/kamailio-ci/pkgs/container/kamailio-ci/34563256…) version: kamailio 6.0.0 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_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 12:27:58 Jan 29 2025 with gcc 14.2.0 It appears that it could have been fixed in 6.0.1 (https://www.kamailio.org/pub/kamailio/6.0.1/ChangeLog commit a5ac9e04b91971da44256373904450501f9a4493) but I don't see 6.0.1-alpine image in https://github.com/kamailio/kamailio-ci/pkgs/container/kamailio-ci. Please advise, is there any plans to publish it? Kind regards, [logo] Arsenijs Kabihno | VoIP SIA "Dynatech", DYNATECH, Jeruzalemes St. 1, Riga, LV-1010, Latvia<https://www.google.com/maps/place/Jeruzalemes+St.+1,+Riga,+LV-1010,+Latvia> a.kabihno(a)dynatech.lv<mailto:a.kabihno@dynatech.lv> | https://dynatech.lv/<https://https://dynatech.lv/> Stay connected: [logo]<https://www.linkedin.com/company/dynatech-latvia/> Confidentiality Notice: This message contains confidential information and is intended only for the named recipient(s). If you are not the addressee you may not copy, distribute or perform any other activities with this information. If you have received this transmission in error, please notify us by e-mail immediately. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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How to remove Reason HF from branched CANCEL?
by Benoit Panizzon 13 May '25

13 May '25

Hi Gang I try to remove the Reason Header from a CANCEL message. I use parallel branching on the registrar. The route below is called very soon in request_route. The cancel matches as I get the log entry just before calling remove_hf("Reason") But the Reason HF is still present when the CANCEL is forked to the registered CPE. The INVITE is routed through a branch_route after forking where headers can be altered per branch. But I don't see the CANCEL in the branch route. Is there a way to alter header in a branched cancel message? # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(FILTER_CANCEL); route(RELAY); } exit; } route[FILTER_CANCEL] { # Buggy iPhones send wrong cancel cause code 200. if (is_present_hf("Reason")) { $var(cause) = $(hdr(Reason){param.value,cause}); $var(text) = $(hdr(Reason){param.value,text}); if (($var(cause) == 200) && ($var(text) == "Call Rejected By User")) { if ($avp(debug) > 0) { xlog("L_INFO", "$cfg(route): Remove wrong iPhone Cancel Reason HF\n"); } remove_hf("Reason"); } } } 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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xhttp module generates an error sip parsing error log
by 崔天浩 13 May '25

13 May '25

### Description I see that when the xhttp module is enabled, kamailio will forge the Via header to process the HTTP request, but the parse sip error will be generated in the log ``` 104(113) ERROR: <core> [core/parser/parse_uri.c:1417]: _parse_ruri(): bad uri </metrics> 104(113) ERROR: <core> [core/parser/parse_uri.c:1455]: parse_orig_ruri(): parse orig ruri failed 104(113) ERROR: pv [pv_core.c:355]: pv_get_ouri_attr(): failed to parse the R-URI ``` ### Troubleshooting #### Debugging Data Set up a Kamailio server with the xhttp module enabled. ``` loadmodule "xhttp.so" event_route[xhttp:request] { $var(xhttp_prom_root) = $(hu{s.substr,0,8}); if ($var(xhttp_prom_root) == "/metrics") { xlogl("L_INFO", " [$utimef(20%y-%d-%m %H:%M:%S)] [$(hdr(Call-ID))]: [bizCall=$xavu(bizCallId)] [bot=$xavu(bizBotId)] Called metrics\n"); prom_dispatch(); } } ``` #### Log Messages ``` 104(113) ERROR: <core> [core/parser/parse_uri.c:1417]: _parse_ruri(): bad uri </metrics> 104(113) ERROR: <core> [core/parser/parse_uri.c:1455]: parse_orig_ruri(): parse orig ruri failed 104(113) ERROR: pv [pv_core.c:355]: pv_get_ouri_attr(): failed to parse the R-URI ``` ### Additional Information ``` root@kamailio-60-bot-67d564c54c-xj97x:/# kamailio -V version: kamailio 6.0.1 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_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 8.3.0 ``` * **Operating System**: ``` root@kamailio-60-bot-67d564c54c-xj97x:/# uname -a Linux kamailio-60-bot-67d564c54c-xj97x 5.4.119-19.0009.28 #1 SMP Thu May 18 10:37:10 CST 2023 x86_64 GNU/Linux ```

1 0

Is there an mistake in the dialog documentation?
by 13715209697＠139.com 12 May '25

12 May '25

Hi, team https://github.com/kamailio/kamailio/blob/master/src/modules/dialog/doc/dia… `dlg.list_match furi eq sip:alice@test.com 2` ? `dlg.list_match furi eq sip:alice@test.com` ?

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Analysis regarding performance regressions OpenSSL 3.0/3.x
by Henning Westerholt 11 May '25

11 May '25

Hello, If somebody is using Kamailio in a larger environment with TLS, this new technical analysis and performance report from haproxy could be interesting: https://www.haproxy.com/blog/state-of-ssl-stacks The bottom-line - OpenSSL 3.0 will show serious performance regressions for larger TLS services with higher performance requirements. OpenSSL 3.1 and newer versions are a bit better, but still much slower in key operations. You should consider using the tls_wolfssl module or stay on OpenSSL 1.1.1. For the medium- to long-term we probably should observe how other OpenSSL libraries are developing and act accordingly for the tls modules. The haproxy project recommends besides using wolfssl also the aws-lc library. The situation regarding OpenSSL 3.x seems to be not easily fixable, as these regressions are caused from internal design decisions. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

The index indicates that the sipjson module is not present in 6.0.x.
by 13715209697＠139.com 10 May '25

10 May '25

https://www.kamailio.org/docs/modules/6.0.x/modules/

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SIPP UAS scenario
by Sergio Charrua 09 May '25

09 May '25

Hi all! my DEV/LAB environment is comprised of 3 nodes: - SIP 1 with Asterisk and SIPP installed (IP 10.20.0.5) - SIP 2 with Kamailio 5.8 (IP 10.20.0.4) - SIP 3 with Asterisk and SIPP installed (IP 10.20.0.6) Kamailio is set as a stateful SIP Proxy and seems to be working fine (the configuration is pretty standard, using dispatcher to dispatch to destination, even though there is only 1 destination endpoint -» SIP 3) When establishing a call from SIP 1 using Asterisk, the termination is correctly done on SIP 3, also running Asterisk. Hanging up call on SIP 3 will send the correct BYE command to SIP 1 (of course, Kamailio is always between the dialogs) When establishing a call from SIP 1 using SIPP and UAC scenario, the termination is correctly done on SIP 3 also running SIPP with a UAS scenario. However, hanging up the call on SIP 3 (SIPP scenario has a Pause of 2 seconds then sends a BYE message), the BYE command is not forwarded by Kamailio and SIPP keep sending BYE commands. This is an example of the ACK and BYE command being sent from SIP 3: > ACK sip:10.20.0.6:5060;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP > 10.20.0.4;branch=z9hG4bK680a.47d763c3f0732dbe0fdb64ccbeb3454a.0 > Via: SIP/2.0/UDP 10.20.0.5:5060 > ;received=10.20.0.5;branch=z9hG4bK01175d44;rport=5060 > Max-Forwards: 69 > From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as023237be > To: <sip:1000@10.20.0.4>;tag=20 > Contact: <sip:anonymous@10.20.0.5:5060> > Call-ID: 568f1bbe197039983fc89cc6403b927b@10.20.0.5:5060 > CSeq: 102 ACK > User-Agent: Asterisk PBX 18.17.1 > Content-Length: 0 and then: > BYE sip:anonymous@10.20.0.5:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.6:5060 > Route: <sip:10.20.0.4;lr=on;ftage=46906> > Max-Forwards: 70 > From: sipp <sip:sipp@10.20.0.6:5060>;tag=46906SIPpTag0120 > To: : "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as023237be > Call-ID: 568f1bbe197039983fc89cc6403b927b@10.20.0.5:5060 > Cseq: 2 BYE > Contact: sip:sipp@10.20.0.6:5060 > Content-Length: 0 > Subject: Sending BYE I'm pretty sure that the issue is within the SIPP Scenario(s), and my guess goes to the UAS scenario on SIP 3 that has some issue with the BYE command, but I can't figure why the message is wrong! The goal is to have a Testing environment where we can simulate different call situations. SIPP is a wonderful tool but such a PITA !! Probably i am missing some detail somewhere.... Has anyone done such a testing environment? Can anyone share a SIPP UAS scenario where the BYE command is sent from the UAS (send BYE instead of recv BYE) ? SIPP is launched on SIP 1 with: sipp -sf uac.xml -i 10.20.0.5 -p 5060 10.20.0.4 -r 1 -m 1 SIPP is launched on SIP 3 with: sipp -aa -sf uas5.xml -i 10.20.0.6 -p 5060 Thanks in advance. Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

2 2

Kamailio Crash
by Ben Kaufman 09 May '25

09 May '25

Greetings, all, I have a redirect server doing CNAM lookup (via http_async) and STIR/SHAKEN validation with secsipidx that is crashing every few hours. Kamailio 6.0.1 running in a Debian 12 container on an Amazon Linux 2023 host with ARM CPU. root@0689d4fe5fed:/# kamailio -v version: kamailio 6.0.1 (aarch64/linux) fce50d flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT-NOSMP, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_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: fce50d compiled on 15:50:40 Apr 1 2025 with gcc 12.2.0 Back trace is here, so I think it's a secsipid related problem? Slightly sanitized this output (Removed SIP message specifics in step #42). What should be my next course of action? #0 __pthread_kill_implementation (threadid=281473196830752, signo=signo@entry=6, no_tid=no_tid@entry=0) at ./nptl/pthread_kill.c:44 #1 0x0000ffff95c90ab4 in __pthread_kill_internal (signo=6, threadid=<optimized out>) at ./nptl/pthread_kill.c:78 #2 0x0000ffff95c4a72c in __GI_raise (sig=sig@entry=6) at ../sysdeps/posix/raise.c:26 #3 0x0000ffff95c3747c in __GI_abort () at ./stdlib/abort.c:79 #4 0x0000ffff95c84aac in __libc_message (action=action@entry=do_abort, fmt=fmt@entry=0xffff95d66d28 "%s\n") at ../sysdeps/posix/libc_fatal.c:156 #5 0x0000ffff95c9aebc in malloc_printerr (str=str@entry=0xffff95d62098 "double free or corruption (!prev)") at ./malloc/malloc.c:5660 #6 0x0000ffff95c9cee8 in _int_free (av=0xffff95db0af0 <main_arena>, p=p@entry=0xaaab05dd41b0, have_lock=<optimized out>, have_lock@entry=0) at ./malloc/malloc.c:4587 #7 0x0000ffff95c9f77c in __GI___libc_free (mem=<optimized out>) at ./malloc/malloc.c:3385 #8 0x0000ffff90eb0a0c in w_secsipid_get_url (msg=0xffff92f53030, purl=0xffff92e84e10 "\220M\350\222\377\377", povar=0xffff92e84fd0 "\004") at secsipid_mod.c:961 #9 0x0000aaaaea4fd710 in do_action (h=0xffffff40a610, a=0xffff92e8e9b0, msg=0xffff92f53030) at core/action.c:1133 #10 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40a610, a=0xffff92e8e9b0, msg=0xffff92f53030) at core/action.c:1620 #11 0x0000aaaaea50db44 in run_actions_safe (h=0xffffff40bc50, a=0xffff92e8e9b0, msg=0xffff92f53030) at core/action.c:1683 #12 0x0000aaaaea71a7e0 in rval_get_long (h=0xffffff40bc50, msg=0xffff92f53030, i=0xffffff40ab48, rv=0xffff92e8fc78, cache=0x0) at core/rvalue.c:973 #13 0x0000aaaaea7200f0 in rval_expr_eval_long (h=0xffffff40bc50, msg=0xffff92f53030, res=0xffffff40ab48, rve=0xffff92e8fc70) at core/rvalue.c:1852 #14 0x0000aaaaea720118 in rval_expr_eval_long (h=0xffffff40bc50, msg=0xffff92f53030, res=0xffffff40b080, rve=0xffff92e903b0) at core/rvalue.c:1862 #15 0x0000aaaaea4fd090 in do_action (h=0xffffff40bc50, a=0xffff92e91810, msg=0xffff92f53030) at core/action.c:1099 #16 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40bc50, a=0xffff92e8d690, msg=0xffff92f53030) at core/action.c:1620 #17 0x0000aaaaea4f9400 in do_action (h=0xffffff40bc50, a=0xffff92e82210, msg=0xffff92f53030) at core/action.c:711 #18 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40bc50, a=0xffff92e82210, msg=0xffff92f53030) at core/action.c:1620 #19 0x0000aaaaea50db44 in run_actions_safe (h=0xffffff40daf0, a=0xffff92e82210, msg=0xffff92f53030) at core/action.c:1683 #20 0x0000aaaaea71a7e0 in rval_get_long (h=0xffffff40daf0, msg=0xffff92f53030, i=0xffffff40c188, rv=0xffff92e82378, cache=0x0) at core/rvalue.c:973 #21 0x0000aaaaea7200f0 in rval_expr_eval_long (h=0xffffff40daf0, msg=0xffff92f53030, res=0xffffff40c188, rve=0xffff92e82370) at core/rvalue.c:1852 #22 0x0000aaaaea720118 in rval_expr_eval_long (h=0xffffff40daf0, msg=0xffff92f53030, res=0xffffff40c6c0, rve=0xffff92e82ab0) at core/rvalue.c:1862 #23 0x0000aaaaea4fd090 in do_action (h=0xffffff40daf0, a=0xffff92e83a90, msg=0xffff92f53030) at core/action.c:1099 #24 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40daf0, a=0xffff92e83a90, msg=0xffff92f53030) at core/action.c:1620 #25 0x0000aaaaea4fd5e8 in do_action (h=0xffffff40daf0, a=0xffff92e83bf0, msg=0xffff92f53030) at core/action.c:1119 #26 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40daf0, a=0xffff92e79450, msg=0xffff92f53030) at core/action.c:1620 #27 0x0000aaaaea4f9400 in do_action (h=0xffffff40daf0, a=0xffff92e75710, msg=0xffff92f53030) at core/action.c:711 #28 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40daf0, a=0xffff92e75710, msg=0xffff92f53030) at core/action.c:1620 #29 0x0000aaaaea50db44 in run_actions_safe (h=0xffffff4101f8, a=0xffff92e75710, msg=0xffff92f53030) at core/action.c:1683 #30 0x0000aaaaea71a7e0 in rval_get_long (h=0xffffff4101f8, msg=0xffff92f53030, i=0xffffff40e028, rv=0xffff92e75878, cache=0x0) at core/rvalue.c:973 #31 0x0000aaaaea7200f0 in rval_expr_eval_long (h=0xffffff4101f8, msg=0xffff92f53030, res=0xffffff40e028, rve=0xffff92e75870) at core/rvalue.c:1852 #32 0x0000aaaaea720118 in rval_expr_eval_long (h=0xffffff4101f8, msg=0xffff92f53030, res=0xffffff40e560, rve=0xffff92e75fb0) at core/rvalue.c:1862 #33 0x0000aaaaea4fd090 in do_action (h=0xffffff4101f8, a=0xffff92e76e10, msg=0xffff92f53030) at core/action.c:1099 #34 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92e74500, msg=0xffff92f53030) at core/action.c:1620 #35 0x0000aaaaea5095fc in do_action (h=0xffffff4101f8, a=0xffff92e79110, msg=0xffff92f53030) at core/action.c:1401 #36 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92e71a40, msg=0xffff92f53030) at core/action.c:1620 #37 0x0000aaaaea4f9400 in do_action (h=0xffffff4101f8, a=0xffff92dd8520, msg=0xffff92f53030) at core/action.c:711 #38 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92dd7e30, msg=0xffff92f53030) at core/action.c:1620 #39 0x0000aaaaea4fd5a8 in do_action (h=0xffffff4101f8, a=0xffff92dd8680, msg=0xffff92f53030) at core/action.c:1115 #40 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92dce4c0, msg=0xffff92f53030) at core/action.c:1620 #41 0x0000aaaaea50dbf4 in run_top_route (a=0xffff92dce4c0, msg=0xffff92f53030, c=0x0) at core/action.c:1703 #42 0x0000aaaaea6bfff8 in receive_msg ( buf=0xaaaaeac284b8 <buf> "INVITE sip:2345678901@1.2.3.4 SIP/2.0\r\nRecord-Route: <sip:1.2.3.5;lr=on;ftag=gK045643d5>\r\nRecord-Route: <sip:1.2.3.6;lr=on;ftag=gK045643d5>\r\nVia: SIP/2.0/UDP 1.2.3.5:5060;branch"..., len=2492, rcv_info=0xffffff410a10) at core/receive.c:520 #43 0x0000aaaaea85abe0 in udp_rcv_loop () at core/udp_server.c:770 #44 0x0000aaaaea4e26a8 in main_loop () at main.c:1895 #45 0x0000aaaaea4f1e88 in main (argc=16, argv=0xffffff411178) at main.c:3406 Kaufman Senior Voice Engineer E: bkaufman(a)bcmone.com SIP.US Client Support: 800.566.9810 | SIPTRUNK Client Support: 800.250.6510 | Flowroute Client Support: 855.356.9768 [img]<https://www.sip.us> [img]<https://www.siptrunk.com> [img]<https://www.flowroute.com> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

2 3

Having issues with compilation of evapi module and evapi_async_relay functions
by Steve Lim 02 May '25

02 May '25

Hello, We're trying to compile the evapi module. Tested on an upgraded 5.8.6 install as well as a new 6.0.1 install. tm module is loaded (for async), but our issues seem to run deeper. evapi_relay and evapi_async_relay are both not found by the core loader as per load failure (same pattern for evapi_async_relay, ommited for brevity) : 0(21496) DEBUG: <core> [core/sr_module.c:766]: find_mod_export_record(): export of <evapi_relay> not found (flags 1) 0(21496) DEBUG: <core> [core/sr_module.c:766]: find_mod_export_record(): export of <evapi_relay> not found (flags 0) 0(21496) ERROR: <core> [core/cfg.y:3877]: yyparse(): cfg. parser: failed to find command evapi_relay (params 2) 0(21496) CRITICAL: <core> [core/cfg.y:4018]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 111, column 43: unknown command, missing loadmodule? earlier up, we're able to set the bind_addr parameter, so it appears that kamailio does recognize the module (and would fail anyways if it didn't since it's supposed to be loaded) 0(21496) DEBUG: <core> [core/modparam.c:112]: set_mod_param_regex(): 'evapi' matches module 'evapi' 0(21496) DEBUG: <core> [core/sr_module.c:852]: find_param_export(): found <bind_addr> in module evapi [/usr/local/lib64/kamailio/modules/evapi.so] 0(21496) DEBUG: <core> [core/modparam.c:135]: set_mod_param_regex(): found <bind_addr> in module evapi [/usr/local/lib64/kamailio/modules/evapi.so] quick look with nm shows that evapi_relay is exported, but evapi_async_relay isn't. root@kamailio-devel:~# nm -D /usr/local/lib64/kamailio/modules/evapi.so | grep relay 0000000000011dd7 T _evapi_relay 0000000000013b3f T evapi_relay 0000000000013b63 T evapi_relay_multicast 0000000000013b8d T evapi_relay_unicast What am I missing out here? Why can't I we get to the evapi_relay exported function and why isn't evapi_async_relay even showing up as exported? Thanks!

1 1

WebRTC error NS_ERROR_NET_TIMEOUT
by David Cunningham 01 May '25

01 May '25

Hello, Does anyone know of a reason (besides the obvious like network down) why users might experience a NS_ERROR_NET_TIMEOUT error in their browser when connecting to WebRTC port 8443 on Kamailio (version 5.2.7)? This only happens intermittently, so debugging it is tricky. The number of connections to Kamailio port 8443 peaks at around 4150 daily. The max open files ulimit in Linux is 65535, and in the Kamailio configuration we have "tcp_max_connections=32768" and "tls_max_connections=32000". Would anyone have any suggestions on where the problem might be? Thanks in advance! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

Looking for Documentation/Guidance to Setup Kamailio and RTPProxy to Terminate Encryption for Asterisk
by Joshua Ruehlig 30 Apr '25

30 Apr '25

Hello, I am fairly new to SIP and just recently setup Asterisk for a few purposes. 1) To have my Hikvision doorbell videocall my phone when someone presses it 2) Videocalling between me and my families' phones/tablets 3) Eventually have a home phone setup for POTS calling. I currently have an ObiHai202 that is still working with GoogleVoice, so this not a priority at the moment. I wanted to have encryption for SIP and RTP outside of my network, but wanted this to all be terminated on my OPNsense router. I can install Kamailio and RTPProxy on OPNsense since they are available for FreeBSD, but it looks like Rtpengine is currently not ported. Ideally I wanted most logic to be handled on my Asterisk server, and have Kamailio/RTPProxy do the bare minimum of encrypting/decryption and rewriting headers as needed. ## Diagram of my Targeted Setup ## * Android Phones / Tablets running Linphone || SIPS(TLS) / SRTP || * OPNsense Router ~ WAN_IP(dynamic)=DOMAIN.com / LAN_IP=10.0.1.1 - Kamailio / RTPProxy || SIP(UDP) / RTP || * FreeBSD Application Server ~ LAN_IP=10.0.1.17 / Internal_IP=192.168.10.17 - Asterisk || SIP(UDP) / RTP || * Hikvision Doorbell ~ Internal_IP=192.168.10.100 #### From my reading so far I have a few questions... * To setup Kamailio > Asterisk it looks like I need to setup the dispatcher module; I also I may need to write some routing code. Am I correct, that I will need to use the dispatcher module? Will I need to modify the routing code/logic or is the default usually sufficient? * I read RTPProxy does support SRTP, but I don't see any configuration options related to encryption. I assume this is just enabled in Asterisk/Kamailio? * For Asterisk configuration I believe I just need to define the outbound_proxy. But now I am not sure how I can require SRTP between my Linphone clients and RTPProxy, while the leg between RTPProxy and Asterisk is unencrypted RTP? Thanks for any help that is provided!

1 2

Kamailio TLS Listener Not Binding to Port (5061 or 10061)
by steven.muchwe 29 Apr '25

29 Apr '25

Hello Kamailio Community, I’ve been troubleshooting an issue with Kamailio where the TLS listener fails to bind to any specified ports (5061, 10061, etc.), despite valid configurations and certificates. Here’s a summary of my setup and steps taken: 1. *Environment Details*: - Kamailio version: 5.7.4 - OpenSSL version: 3.0.13 - Operating System: Ubuntu (Noble Release) - TLS module (tls.so) is installed and loaded. 2. *Issue Details*: - Configurations validate successfully (config file ok). - OpenSSL works perfectly when testing certificates and keys with s_server, binding to ports (5061 and others). - Kamailio fails to bind TLS listeners (ss -tulnp shows no activity on the specified ports). 3. *Steps Already Taken*: - Simplified TLS configuration (minimal_tls.cfg) with: plaintext listen=tls:10.14.202.39:5061 loadmodule "tls.so" modparam("tls", "certificate", "/home/localtech/vicissl/868a963bc33d5eae.crt") modparam("tls", "private_key", "/home/localtech/vicissl/private.key") - Tested multiple ports (5061, 10061, 15061). - Checked firewall settings (iptables) and confirmed no restrictions. - Rebuilt Kamailio from source and ensured TLS modules are linked to OpenSSL. - Ran Kamailio with maximum debugging (-ddd) to examine logs—no binding-related errors appeared. 4. *Log Excerpts*: (Attach relevant logs showing TLS initialization or lack of binding activity.) 5. *Question*: What additional steps or configurations should I explore to resolve this issue? Could this be a compatibility problem between Kamailio 5.7.4 and OpenSSL 3.0.13? Any guidance or insights would be greatly appreciated! Best regards, Steven Muchwe Njoroge

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Re: SIP Call Drops After 0.32 Seconds – Asterisk on Kubernetes with Kamailio + RTPengine
by airsay＠duck.com 29 Apr '25

29 Apr '25

I will let the more experienced Kamailio folks comment on the technical/Kamailio/RTP modifications that you need to make. The SIP standard requires an ACK to a 200 OK to be sent within that 32 second window. If an ACK is not received, the session is torn down. You would need to do a packet capture to review the Contact header in the SIP and the c: in the SDP. I'm just getting bedded in with SIP, reading Alan B Johnston's "SIP: Understanding the Session Initiation Protocol". Excellent introduction to understanding SIP. Best regards Sent from my iPhone > On 23 Apr 2025, at 16:35, Fernando Lopes via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello everyone, > I'm running into an issue with SIP calls in my current setup and would really appreciate some help. > > Setup: > I have a machine named sip00 (IP: 192.168.1.75) running Kamailio + RTPengine. > Kamailio is dispatching calls to sip:192.168.1.190:32210;transport=tcp. This IP points to another machine running Asterisk inside a Kubernetes cluster. > RTPengine is configured with an RTP port range of 10000–20000, and my router is set to allow that same range. > > Asterisk Kubernetes Service Configuration: > yaml > Copy > Edit > spec: > ports: > - name: tcp-port > protocol: TCP > port: 5060 > targetPort: 5060 > nodePort: 32210 > - name: udp-port > protocol: UDP > port: 5060 > targetPort: 5060 > nodePort: 32210 > > Problem: > When I initiate a SIP call, the router forwards traffic to Kamailio + RTPengine, which then sends it to the Asterisk server on 192.168.1.190. > Everything seems fine initially, but at exactly 0.32 seconds into the call, Asterisk sends a BYE and no longer responds with 200 OK to the SIP dialog — even though I'm still receiving and sending audio. Then, at around 01:04, I get a 408 Request Timeout. > > Questions: > Do I need to explicitly expose the RTP port range (10000–20000) in the Asterisk Kubernetes service as well? > Why is Asterisk sending a BYE so early if audio is still flowing? > Could it be a signaling timeout or an issue with SIP dialog tracking? > Any help or pointers would be greatly appreciated! > > Thanks in advance! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender!

7 7

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

SIP Call Drops After 0.32 Seconds – Asterisk on Kubernetes with Kamailio + RTPengine
by Fernando Lopes 29 Apr '25

29 Apr '25

Hello everyone, I'm running into an issue with SIP calls in my current setup and would really appreciate some help. Setup: I have a machine named sip00 (IP: 192.168.1.75) running Kamailio + RTPengine. Kamailio is dispatching calls to sip:192.168.1.190:32210;transport=tcp. This IP points to another machine running Asterisk inside a Kubernetes cluster. RTPengine is configured with an RTP port range of 10000–20000, and my router is set to allow that same range. Asterisk Kubernetes Service Configuration: yaml Copy Edit spec: ports: - name: tcp-port protocol: TCP port: 5060 targetPort: 5060 nodePort: 32210 - name: udp-port protocol: UDP port: 5060 targetPort: 5060 nodePort: 32210 Problem: When I initiate a SIP call, the router forwards traffic to Kamailio + RTPengine, which then sends it to the Asterisk server on 192.168.1.190. Everything seems fine initially, but at exactly 0.32 seconds into the call, Asterisk sends a BYE and no longer responds with 200 OK to the SIP dialog — even though I'm still receiving and sending audio. Then, at around 01:04, I get a 408 Request Timeout. Questions: Do I need to explicitly expose the RTP port range (10000–20000) in the Asterisk Kubernetes service as well? Why is Asterisk sending a BYE so early if audio is still flowing? Could it be a signaling timeout or an issue with SIP dialog tracking? Any help or pointers would be greatly appreciated! Thanks in advance!

2 1

rtpengine module does not detect new nodes in table
by David Cunningham 29 Apr '25

29 Apr '25

Hello, We are using Kamailio 5.8.2 with the rtpengine module on Ubuntu 22.04. The "rtpengine" table is used to define nodes, with a configuration like this: modparam( "rtpengine", "db_url", DB_URL ) modparam( "rtpengine", "table_name", "rtpengine" ) modparam( "rtpengine", "setid_default", 1 ) It works fine if Kamailio is restarted after entries in the "rtpengine" table are inserted or deleted, but new entries in the table aren't detected automatically. Is there a way to configure Kamailio to detect them? Thank you, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

4 6

OT: Client DNS resolver crashing on NAPTR lookup
by Benoit Panizzon 28 Apr '25

28 Apr '25

Hi List I hope it is OK to post a slightly off topic question. I have not found any similar reports via google. During testing of our development platform, I found two independent clients whose resolver process crashes randomly, when performing NAPTR lookups. Mitel/Aastra 68XXi Phones with Firmware >= 6.4 up to most recent. Inalp/Patton SNXX (Trinity OS) SIP Gateway with Firmware >= 3.14.X up to most recent. Both vendors reply they have no prior reports of this issue. The hostname used as registrar / proxy points to a NAPTR entry, which in turn points to 3 SRV entries für SIP+D2T, SIP+D2U, SIPS+D2T Each SRV entry then points to 4 IP Addresses (2x ipv4 and 2x ipv6) So quite a complex lookup and quite large replies. Anyone else observed this issue? When using a hostname that merely points to the A and AAAA records, no crashes. 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

Kamailio Redis integration
by Timothy Delin 28 Apr '25

28 Apr '25

Hey team, I am facing an issue where my Kamailio and Redis are not integrated. I am using a VM with RHEL 9 OS, Kamailio in 5.4.7 version, and Redis DB in 7.2.7 version. Both my Kamailio and Redia are using Docker. We have installed the libhiredis and the ndb_redis in Kamailio When we tried to do a simple print "Hello World" from the Redis in Kamailio, we are receiving this error "ERROR: could not open module, missing loadmodule" even though we have confirmed in kamcmd Best regards, Timothy

3 2

Authentication qop auth-int
by 13715209697＠139.com 28 Apr '25

28 Apr '25

Hello! Kamailio 5.7.6 Debian 10.12 Here is my authentication route: ``` route[AUTH] { if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "2"); # flag not 0 exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } ... } ``` start eyebeam Version 1.5.14.4 ``` eyebeam --->REGISTER ----> kamailio <--401 (WWW-Authenticate: Digest realm="192.168.3.10", nonce=..., qop="auth-int") --->REGISTER (Authorization: Digest username="08001",realm="192.168.3.10",nonce=...,uri=...,response=...,cnonce=...,nc=00000001,qop=auth-int,algorithm=MD5) <--401 (auth_challenge() return -2 ) ``` OpenSIPS 3.4.12 register to kamailio, got the same result as eyebeam Any suggestions? Thanks!

4 8

Kamailio v6.0.1 and SUBSCRIBE processing
by Blair, Steve 26 Apr '25

26 Apr '25

I’m coming back to Kamailio after about a 10 year absence. I have v6.0.1 running handling registrations (unique devices) and inbound/outbound PSTN calling correctly. What I need is to process subscribe requests. I’m looking for direction on the correct configuration to 1. Authenticate subscribe messages. This appears to be working with following snippet. #eDevice: Need to authenticate subscribes before proceeding xlog("L_INFO", "eDevice: Start initial SUBSCRibe check. \n"); if (is_method("SUBSCRIBE")) { record_route(); if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); } # eDevice SUB Change consume_credentials(); route(PRESENCE); } 1. After authenticating forward any subscribe to an external presence server 2. Relay any response from external presence server to client device. Only the above authentication is working so direction on proper processing of subscribes when Kamailio is not the presence server will be appreciated. Thanks

1 0

Re: RTPengine media starts for intiating caller only
by airsay＠duck.com 23 Apr '25

23 Apr '25

Again, not super skilled yet with Kamailio/RTPEngine, but my first instinct is: "why you are advertising myproxyname.domain.net which is possibly a public IP". Your entire setup is in your LAN so you don't need to advertise any public IP. Look forward to others contributing and clarifying that Sent from my iPhone > On 23 Apr 2025, at 20:42, smh2017--- via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello, > > I have a problem with my basic Kamailio setup using RTPengine. I'm testing initially with two client UAs in the same LAN as the SIP server behind the firewall. The initiating UA starts the media proxy as soon as the receiving UA accepts the call, indicated by the call timer starting. But after accepting the call the receiving UA's timer doesn't start and after 10-15 secs hangs up with a NO ACK error message. > > I am using the kamailio default config file that comes with the Debian package and have made minimal changes recommended by various How-Tos for using Kamailio behind a firewall. Because the clients can successfully register over TLS, I'm assuming the problem is in the rtpengine configuration. The two rtp listen parameters I've tried are: > > interface = internal/192.168.1.100;external/192.168.1.100!myproxyname.domain.net > --or-- > interface = internal/192.168.1.100;external/myproxyname.domain.net > listen-ng = localhost:2223 > > I'm not sure why some example rtpengine configs list both the private and public addresses as external. > > Any suggestions where to start troubleshooting? > > Thanks, > > Steve > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender!

2 1

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