- sr-users - kamailio.org

siptrace module not respecting send_sock after kamailio 5.6.6
by Daniel Donoghue 27 May '25

27 May '25

Hello, I have been using the siptrace module on a multihomed kamailio setup for some time now, with kamailio 5.5.7 and 5.6.6, without incident. I am listening to my external interface using something like: listen=tls:external_ip:5061 listen=tcp:external_ip:5060 Listen=tcp:internal_ip:5000 name "homer" And for siptrace... modparam("siptrace", "duplicate_uri", "sip:homer_ip:9060;transport=tcp") modparam("siptrace", "send_sock_name", "homer") I have also used send_sock_addr and force_send... This works as expected. I recently deployed this same setup using kamailio 6.0.1 and discovered that the send_sock is not being respected, even though the logs say it has been selected. Instead, tcpdump and careful examination of all the file descriptors for the kamailio processes reveal it is sending with the source set to my external_ip instead of the internal_ip. I have verified this on the same server, ruling out os level issues routing issues and have tried manually connecting using nc with the internal ip address for the source, which works. I tried downgrading, and found that the same is true for kamailio 5.7.6, 5.8.6 and 6.0.1 This issue only appears to affect siptrace as other sockets appear to use the correct source ip to, for example, connect to a database. I had a look at the change logs and documentation and cant see anything obvious that I need to change from a configuration perspective. Could anybody help? I would prefer to be running 6.0+ in my new production environment but wont be able to if this cannot be solved quickly. Many thanks for your help. With every blessing, -- Daniel Donoghue

1 0

Kamailio Regex
by Timothy Delin 26 May '25

26 May '25

Hi everyone, I need your advice. I am extracting a SIP Body message from a User Agent Client using the regex as such "$var(calling_party)= $(var(body){re.subst,/.*Calling-party:\s*sip:(\+[0-9]+)@.*/\1/});" And I tried to log it as "xlog("L_INFO", "Extracted number 1: $var(calling_party) <><><>\n");" However, the output in my logs is only showing as below, where I cannot see any value after my variables. "1(6) INFO: {1 10 SUBSCRIBE 5017ae21204c72a3} <script>: Extracted number 1: +6512345678" I suspect this is the reason why when I tried to use this variable in a redis_cmd, it returns as null as the variable may have impacted the command input to Redis. I tried to change the $var(calling_party)="+6512345678" and it is working perfectly fine Best regards, Timothy

3 4

How to count the first re-register request?
by James Morrison 26 May '25

26 May '25

I want to count the first REGISTER request during an initial register or re-register from UE. In my scscf instance, I have defined att_init_reg and att_rereg counters to count the number of initial registers and re-registers respectively. In my early tests, the following piece of logic worked as expected to filter the very first register request (one that is sent from UE before getting challenged). However, more tests showed that during re-registers, UE sends a REGISTER request, containing an authorization header and non-empty response, in that case the code is unable to distinguish a re-register attempt and the related counter fails to increment. Is there a way to fix this issue? #!ifdef WITH_XHTTP_PROM /* Only count the first register request */ if (!is_present_hf("Authorization") || search_hf("Authorization","response=\"\"","f")) { if ($var(impu_registered) != 1) { prom_counter_inc("att_init_reg", "1"); } else { prom_counter_inc("att_rereg", "1"); } } #!endif

3 5

5.7.7 release, tag and docker build
by Henning Westerholt 23 May '25

23 May '25

Hello, I have noticed that the git tag for release 5.7.7 was missing, I've just set it. The 5.7.7-bullseye docker image is also missing from https://github.com/kamailio/kamailio-docker/pkgs/container/kamailio Not sure if this is done automatically, if not - please can somebody please trigger the workflow to create it? Thank you, Henning

1 0

Help Adding WebRTC to Existing SIP Setup
by Fernando Lopes 22 May '25

22 May '25

Hi everyone, I currently have a working SIP setup, but I'm trying to get WebRTC working alongside it and haven’t had much success so far. Does anyone have tips or best practices for enabling both SIP and WebRTC in the same configuration? Is it possible to handle both within a single Kamailio config? Or would I need some kind of conditional logic (like if/else) to manage how the config handles different types of clients? My setup looks like this: Router → Kamailio → Kubernetes (Asterisk inside the cluster) I was having some issues with the Contact header, but I managed to resolve that. Now I'm stuck figuring out how to make WebRTC work cleanly alongside SIP. Any advice would be appreciated! Thanks!

2 1

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

21 May '25

Hello, Kamailio SIP Server v5.7.7 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.7. Note that 5.7 is now the third last stable branch, v5.7.7 being the last planned released in 5.7.x series. The latest two stable branch are 5.8 and 6.0, with v6.0.1 being released a while ago. For more details about version 5.7.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/05/kamailio-v5-7-7-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 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

1 1

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

1 0

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

4 6

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)

1 1

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.

2 1

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 ______________________________________________________

2 2

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` ?

2 2

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/

3 2

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

Kamailio Asterisk Realtime integration
by cedric＠mrtn.ovh 02 May '25

02 May '25

Hello, I'm trying to use Kamailio as proxy for Asterisk. I follow this tutorial https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I maked some updates to replace old libs and old elements related to db table and fields to use the pjsip table (ps_auths) and fields (username, password). I configured Asterisk Realtime by following the Asterisk documentation (extconfig.conf, sorcery.conf, rec_odbc.conf etc...). I'm lock at this step : The registering is handle correctly by Kamailio and forwarded to Asterisk, but asterisk respond with a "401" error and want to get a digest auth... Bellow, my kamailio config file: ``` #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_DEBUG # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asterisk_password@localhost/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.1.2:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.2" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "192.168.1.2" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "jsonrpcs.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs fifo params ----- modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "ps_auths", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif ``` Also, I'm not sure about the configuration to add into asterisk, like a kamailio registration block or endpoint block, and... probably others... Thanks to help. Flywall.

1 0

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

4 8

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

RTPengine media starts for intiating caller only
by smh2017＠zoho.com 23 Apr '25

23 Apr '25

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

1 0

404 sent from Kamailio NOT from the script
by Richard Robson 23 Apr '25

23 Apr '25

I have a problem where a call comes in and causing a 404 Not Found to be replied. the script does not have a 404 reply set in it anywhere unfortunately I cannot get the script to produce logs, hence my other post. But its making it tricky to find the source of the 404. following the script it should not be producing a 404, but doing a relay. I suspect there is something wrong with the tm or its not getting a result it needs. It looks like the transaction module is triggering thge 404 somehow and Its showing up in debug below. Does anyone have any ideas why i'm getting a 404? version 5.6.6 Regards, Richard Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:1100]: t_check_msg(): msg (0x7faf977b64b0) id=4/28700 global id=3/28700 T start=0xffffffffffffffff Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:503]: t_lookup_request(): start searching: hash=60776, isACK=0 Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:463]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK-2e6a90ab-1bfb9be-d4ed8e5d-7f0300025430] Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:703]: t_lookup_request(): no transaction found Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:1172]: t_check_msg(): msg (0x7faf977b64b0) id=4/28700 global id=4/28700 T end=(nil) Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:1394]: t_newtran(): msg (0x7faf977b64b0) id=4/28700 global id=4/28700 T start=(nil) Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:503]: t_lookup_request(): start searching: hash=60776, isACK=0 Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:463]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK-2e6a90ab-1bfb9be-d4ed8e5d-7f0300025430] Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:703]: t_lookup_request(): no transaction found Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} <core> [core/mem/q_malloc.c:385]: qm_malloc(): qm_malloc(0x7faf9301f000, 8336) called from tm: h_table.c: build_cell(336) Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_hooks.c:342]: run_reqin_callbacks_internal(): trans=0x7faf932f42c0, callback type 1, id 0 entered Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} <core> [core/mem/q_malloc.c:385]: qm_malloc(): qm_malloc(0x7faf9301f000, 40) called from tm: t_hooks.c: insert_tmcb(134) Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:1100]: t_check_msg(): msg (0x7faf977b64b0) id=4/28700 global id=4/28700 T start=0x7faf932f42c0 Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_lookup.c:1176]: t_check_msg(): T (0x7faf932f42c0) already found for msg (0x7faf977b64b0)! Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} <core> [core/mem/q_malloc.c:385]: qm_malloc(): qm_malloc(0x7faf9301f000, 626) called from tm: t_reply.c: _reply_light(488) Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_reply.c:1768]: cleanup_uac_timers(): RETR/FR timers reset Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_hooks.c:252]: run_trans_callbacks_internal(): DBG: trans=0x7faf932f42c0, callback type 512, id 0 entered Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7faf932f42c0) event type 512, reply code 404 Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_reply.c:625]: _reply_light(): reply sent out - buf=0x7faf977bf7f8: SIP/2.0 404 Not Foun... shmem=0x7faf932f0468: SIP/2.0 404 Not Foun Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_funcs.c:122]: put_on_wait(): put T [0x7faf932f42c0] on wait Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} <core> [core/mem/q_malloc.c:495]: qm_free(): qm_free(0x7faf976b5010, 0x7faf977bf7f8), called from tm: t_reply.c: _reply_light(635) Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} tm [t_reply.c:636]: _reply_light(): finished Apr 22 14:50:51 kamai01 kamailio[28700]: DEBUG: {1 1 INVITE afc0c04-9d8bc3d-1bfb9be-7f02f033b5f0-2406050a-13d8-7225} sl [sl.c:288]: send_reply(): reply in stateful mode (tm)

4 7

Kamailio docker with missing modules
by Martin Nyström 23 Apr '25

23 Apr '25

Hi, I’m using the docker ghcr.io/kamailio/kamailio:5.8.6-xenial but it appears to be missing the app_python modules. How can I add it, together with other missing modules? /M

1 2

Behaviour of suspended transactions
by Alex Balashov 18 Apr '25

18 Apr '25

Hello! Two questions about suspended transactions: 1) What is the lifetime of a suspended INVITE transaction? Can I change it? 2) While a transaction is suspended, its event hooks (e.g. onreply_route, failure_route) do not run, of course, even if replies that match the transaction are received. Is there a way to intercept these replies externally and resume the transaction upon the receipt of the reply, and prior to the processing of the reply? For example, can the replies be captured in a stateless route, and mapped Call-ID => index/label to t_continue() the transaction, then have the transaction state machine process that very same reply too? t_is_retr_async_reply() seems to hint at this intention, but its scope is more narrow, if I understood the documentation correctly: it seems to deal with retransmissions only. The code path I'm interested in is more akin to a t_is_reply_to_async() function. As you might guess, the driver for this inquiry is a mobile workflow where, in the vast majority of cases, one is contacting a stale registration and should just suspend the transaction, but where occasionally, in a small minority of cases, the endpoint does respond on the existing contact. The standard tsilo workflow of trickling parallel branches into a pending transaction doesn't work here because of the behaviour of Apple's CallKit; we have to try the existing contact _before_ we send a push notification. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

3 2

XLOG not writing to local0
by Richard Robson 17 Apr '25

17 Apr '25

I have setup the kamailio.log file to receive the logs from kamailio and I am receiving some, but I do mot see any logs from xlog commands in the script. I see logs like this on startup and various others, but none from xlog no matter how I format them. can anyone advise where I may be going wrong? current logs: Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9073]: INFO: <core> [core/udp_server.c:163]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 for fd 12 Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9073]: INFO: <core> [core/udp_server.c:233]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 on fd 12 Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9073]: INFO: <core> [core/udp_server.c:163]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 for fd 13 Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9073]: INFO: <core> [core/udp_server.c:233]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 on fd 13 Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9073]: INFO: <core> [core/udp_server.c:163]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 for fd 14 Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9073]: INFO: <core> [core/udp_server.c:233]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 on fd 14 Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9118]: INFO: jsonrpcs [jsonrpcs_sock.c:453]: jsonrpc_dgram_process(): a new child 0/9118 Apr 17 12:28:49 server01 /usr/local/sbin/kamailio[9120]: INFO: ctl [io_listener.c:217]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) config: debug=4 log_stderror=no log_facility=LOG_LOCAL0 log_prefix="[$mt $hdr(CSeq) $ci] " example xlogs that do not print (they are the start of request route): xlog("LOG_LOCAL0","L_ERR","RIK: Start route\n"); xlog("TRACE: Start Route [Main] for [Call-ID: $ci]\n");

2 2

IPv6 bind wrong IP. #4199
by Vu Nguyen 17 Apr '25

17 Apr '25

*Dear Sir,* I have opened an issue on GitHub: *"IPv6 bind wrong IP. #4199 <https://github.com/kamailio/kamailio/issues/4199>"*. *System Details:* - *Kamailio (5.8.x)* – SIP Proxy - *FreeSWITCH* – Media Server - *OS* – Debian 12 - *Gateway* – aaaa:bbb:cccc:20::1 - *IPv6 Address* – aaaa:bbb:cccc:20::41e ------------------------------ *kamailio.cfg:* #!define ADDR_IPV4 xxx.xxx.xxx.xxx #!define ADDR_IPV6 [aaaa:bbb:cccc:20::41e] listen=ADDR_IPV4 listen=ADDR_IPV6 ------------------------------ *netstat Output:* tcp6 0 0 aaaa:bbb:cccc:20:::5060 :::* LISTEN 1691274/kamailio tcp6 0 0 aaaa:bbb:cccc:20:::5061 :::* LISTEN 1691274/kamailio udp6 0 0 aaaa:bbb:cccc:20:::5060 :::* 1691222/kamailio As shown, although I explicitly set the listen IP to aaaa:bbb:cccc:20::41e, Kamailio strips the last part of the address and binds to aaaa:bbb:cccc:20:: I also upgraded to *version 6.0* with default cfg and set auto_bind_ipv6=1 or changed ipv6 address, but the issue still persists. ------------------------------ *IPv6 Interface Configuration:* auto eth1 iface eth1 inet6 static address aaaa:bbb:cccc:20::41e netmask 64 gateway aaaa:bbb:cccc:20::1 I can successfully ping in and out using this IPv6 address. ------------------------------ Do I need to configure anything else in Kamailio or the system to ensure it binds to the full IPv6 address aaaa:bbb:cccc:20::41e as expected? Currently I can register with ipv4, but ipv6 does not. Thank you for your assistance.

3 2

Handling NAT Keepalive in IMS PCSCF Configuration: Bypassing NATHelper Module
by f.soltanzade72＠gmail.com 16 Apr '25

16 Apr '25

Hi All, In the default Kamailio configuration, keepalive handling for NAT is typically managed through the NATHelper module. However, in the IMS PCSCF configuration, the NATPING route is used to send SIP OPTIONS messages for keepalive, which bypasses the NATHelper module. This approach makes it impossible to send raw UDP packets for keepalive handling. Can any one help me configuring that? Thank you.

1 0

How to place a delay between cancelling branch and retrying next one on timer expiry?
by Alex A 15 Apr '25

15 Apr '25

Hi All, I have an interesting use-case where I need to place a delay between cancelling branch and retrying next one. *Background: * Running Kamailio 5.6.x Kamailio routes requests via a freeswitch B2BUA server on select call flows (where B2BUA functionality is required). Call flow examples: PSTN --> Kamailio --> Freeswitch --> appserver1 PSTN --> Kamailio --> Freeswitch --> appserver2 On attempt failures, a branch retry is required from appserver1 to appserver2; however freeswitch B2BUA remains the same Since Freeswitch does not quite follow the RFC 3261 timer K specifications (should be 0 seconds for reliable transports, but FS keeps transactions for T4 (set to 2 seconds)); I have implemented an artificial delay in failure route before retrying (as per recommendation described here: https://freeswitch-users.freeswitch.narkive.com/TsSye66D/482-request-merged… ) *Script config: * onreply_route[MANAGE_REPLY] { xlog("L_INFO", "reply $T_reply_code received"); if (t_check_status( "1[1-9][0-9]")) { t_reset_fr(); } else { t_set_fr(120000); } } route(NATDETECT); #!ifdef WITH_RTPENGINE route(RTPMANAGE); #!endif } route[RELAY] { if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); } if (!t_relay()) { sl_reply_error(); } exit; } route[TOCARRIER] { process bunch of things, attach custom headers, generate request URI t_on_failure("MANAGE_FAILURE_CARRIER"); route(RELAY); } failure_route[MANAGE_FAILURE_CARRIER] { xlog("L_INFO", "reply $T_reply_code received"); if ($avp(inbound_attempt)<=1){ if ($dlg_var(b2bua)=="true"){ usleep(1900000); } route("TOCARRIER"); } *Call Flows: * The above fix works well when an explicit error reply is received back. *error reply SIP exchange: * SBC --> INVITE(branch0) ---> Freeswitch SBC <-- 503 <-- Freeswitch failure_route executed (sleep runs here) SBC --> INVITE(branch1) ---> Freeswitch call continues as expected. however does not work when NO reply is received at all and the timer expires. *Current timer expiry SIP exchange: * SBC --> INVITE(branch0) ---> Freeswitch SBC <-- 100/180 <-- Freeswitch SBC (timer expires) failure_route executed (sleep runs here) SBC --> INVITE(branch1) ---> Freeswitch SBC <-- 482 Merged <-- Freeswitch SBC --> CANCEL(branch0) ---> Freeswitch I am looking to CANCEL branch0 before the sleep delay runs Based on source code, t_cancel_branches("this") can only be executed in reply_route; however the chellenge is that reply_route never sees the "faked 408" on timer expiry(although the failure route does see $T_reply_code 408). Any suggestions on how to handle this would be greatly appreciated. TIA

6 11

RTPEngine pool selection for in-dialog requests
by Ben Kaufman 15 Apr '25

15 Apr '25

I'm looking for information on the correct way to handle calls to RTPEngine when processing in-dialog requests. I currently have several setid's configured for varying usages. During processing of an initial INVITE, a determination is made as to if the call will require the use of RTPEngine, and if so, which of the sets should be used. The following is done in code: * setid is assigned to AVP $avp(rtpengine_pool_num) * set_rtpengine_set("$avp(rtpengine_pool_num)") is called, * rtpengine_offer is then called. * Additionally, a record-route parameter of proxy_media=yes is added. * The flag PROXY_MEDIA is set In the reply route, the PROXY_MEDIA flag is checked, and if set, set_rtpengine_set("$avp(rtpengine_pool_num)") is called, followed by rtpengine_answer. When handling an in-dialog message my code didn't have anything to evaluate for the setid. rtpengine_offer was just called. Looking at the debug from the rtpengine module, I can see that the record is found in the hashtable, however the node indicated in that result is not necessarily used - rtpengine will select an entry from the default setid instead. From the rtpengine module logs, IP addresses redacted: 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine_funcs.c:139]: check_content_type(): type <application/sdp> found valid 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:2249]: build_rtpp_socks(): same rtpengines list version: 1 (1744651613) 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:3818]: select_rtpp_node_old(): rtpengine hash table lookup find node=udp:A.B.C.D:1234 for calllen=36 callid=ac66f56b-93f8-123e-1b99-0242ac120002 viabranch= 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:3702]: select_rtpp_node_new(): sum is = 210 2(9) DEBUG: {time=1744651756.968228} rtpengine [rtpengine.c:3198]: rtpp_function_call(): proxy reply: d3:sdp288:v=0 o=FreeSWITCH 1744611570 1744611571 IN IP4 W.X.Y.Z s=FreeSWITCH c=IN IP4 W.X.Y.Z My question is really: **SHOULD** rtpengine_offer (and rtpengine_manage, etc) use the node that it finds in the hash table lookup automatically, or should I have to explicitly set the setid? For now I've solved the problem by adding the setid to the record-route headers so I can easily retrieve it in dialog, but it seems that if the module finds a record for a node that there should be some way to rely on that explicitly. 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 2

Thank you Guys
by Fernando Lopes 15 Apr '25

15 Apr '25

Just want to say thank you for the support. I spent two weeks working on a config, and even though it didn’t work out, I really appreciate the learning process. Today, I found a config that worked here: 🔗 https://github.com/altanai/kamailioexamples/tree/master/RTPEngine_media_pro…

1 0

Nat Problems Kamailio
by Fernando Lopes 14 Apr '25

14 Apr '25

Hello Everyone, I'm having some trouble getting my SIP setup to work properly. 📞 My Call Flow: SIP Client initiates a call The call reaches my router, which forwards it to Kamailio + RTPengine running on 192.168.1.75:5060, which then forwards it to A Kubernetes machine at 192.168.1.190:32210, where Asterisk is running 🔧 Network Setup: Router has the following ports forwarded: UDP 5060 UDP 10000–20000 (for RTP) STUN is enabled on the SIP client ✅ What Works: When I’m connected through a VPN (i.e., inside the local network), everything works fine — both incoming and outgoing audio. ❌ The Problem: When calling from outside the network, signaling works and the call connects, but I can’t receive or send any audio. Here is a little of my config of kamailio modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "setid_avp", "$avp(setid)") listen=udp:192.168.1.75:5060 advertise 62.2x.xxx.xxx:5060 listen=tcp:192.168.1.75:5060 advertise 62.2x.xxx.xxx:5060 listen=udp:127.0.0.1:5060 #Nat Helper modparam("nathelper", "force_socket", "192.168.1.75:5060") modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "nat_addr_mode", 1) request_route{ ... route(init); ... if(is_method("INVITE|ACK|BYE|CANCEL|OPTIONS|INFO|UPDATE|PRACK"))\{ route(handle_sip); exit; } } route[init]{ ... if (nat_uac_test("19")){ setflag(FLT_NAT); fix_nated_contact(); force_rport(); } } route[handle_sip]{ if ($rm == "INVITE"){ if (isflagset(FLT_NAT)){ route(natmanage); } if (!ds_select_dst("1", "4")){ sl_send_reply("500", "No destination"); exit; } $ru = "sip:" + $rU + "@" + $rd; # Keeps sip:100@domain $du = "sip:192.168.1.190:32210"; # Matches your dispatcher list route(relay); } if ($rm == "BYE" || $rm == "CANCEL"){ if (isflagset(FLT_NAT)){ rtpengine_delete(); } route(relay); } if ($rm == "ACK"){ if (isflagset(FLT_NAT)){ route(natmanage); } route(relay); } route(relay); } route[natmanage]{ if (is_request()){ if (has_body("application/sdp")){ rtpengine_offer("replace-origin replace-session-connection ICE=remove"); } force_rport(); fix_nated_contact(); }else{ rtpengine_answer("replace-origin replace-session-connection ICE=remove"); } } rtpengine.conf [rtpengine] table = 0 interface = internal/192.168.1.75 listen-ng = 127.0.0.1:2223 log-level = 7 timeout = 60 silent-timeout = 3600 tos = 184 port-min = 10000 port-max = 40000 And here are some logs when I call from outside 2025/04/13 15:58:33.407265 2.8x.xx.xxx:59396 -> 192.168.1.75:5060 INVITE sip:100@192.168.1.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:59396;rport;branch=z9hG4bKPj92490d6b6d0c4cd88dac9f0043b1642e Max-Forwards: 70 From: "6001" <sip:6005@192.168.1.75:5060>;tag=0397d2bdc3ef47f5b6e498126a360b0f To: <sip:100@192.168.1.75:5060> Contact: "6001" <sip:6005@192.168.1.116:59396;ob>;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8963 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Content-Type: application/sdp Content-Length: 537 v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.116 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4032 RTP/AVP 8 0 101 c=IN IP4 192.168.1.116 b=TIAS:64000 a=rtcp:4017 IN IP4 192.168.1.116 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1567952351 cname:73284f1717045a81 a=ice-ufrag:37e51dc0 a=ice-pwd:49f7442b5078148140877b44 a=candidate:Hc0a80174 1 UDP 2130706431 192.168.1.116 4032 typ host a=candidate:Hc0a80174 2 UDP 2130706430 192.168.1.116 4017 typ host 2025/04/13 15:58:33.412683 192.168.1.75:5060 -> 192.168.1.190:32210 INVITE sip:100@192.168.1.75 SIP/2.0 Via: SIP/2.0/UDP 62.2x.2xx.xxx:5060;branch=z9hG4bK6362.26a75297a4ccf4984dd1f68644c2148f.0 Via: SIP/2.0/UDP 192.168.1.116:59396;received=2.8x.xx.xxx;rport=59396;branch=z9hG4bKPj92490d6b6d0c4cd88dac9f0043b1642e Max-Forwards: 69 From: "6001" <sip:6005@192.168.1.75:5060>;tag=0397d2bdc3ef47f5b6e498126a360b0f To: <sip:100@192.168.1.75:5060> Contact: "6001" <sip:6005@2.8x.xx.xxx:59396;ob>;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8963 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Content-Type: application/sdp Content-Length: 322 v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.75 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 12156 RTP/AVP 8 0 101 c=IN IP4 192.168.1.75 b=TIAS:64000 a=ssrc:1567952351 cname:73284f1717045a81 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:12157 2025/04/13 15:58:33.428169 2.8x.xx.xxx:59396 -> 192.168.1.75:5060 INVITE sip:100@192.168.1.75:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:59396;rport;branch=z9hG4bKPj14c6f24b814e4d158014cc80b1fa5f8f Max-Forwards: 70 From: "6001" <sip:6005@192.168.1.75:5060>;tag=0397d2bdc3ef47f5b6e498126a360b0f To: <sip:100@192.168.1.75:5060> Contact: "6001" <sip:6005@192.168.1.116:59396;ob>;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8964 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Authorization: Digest username="6005", realm="asterisk", nonce="1744559913/dab8894e3f31e038d7a969a6349a11c2", uri="sip:100@62.2x.2xx.xxx", response="2ce500d6a1583f05f6344bdf1f712fc7", algorithm=MD5, cnonce="808fc766135d40589dd68032bcaab5c6", opaque="2232aa9c42080f12", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 537 v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.116 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4032 RTP/AVP 8 0 101 c=IN IP4 192.168.1.116 b=TIAS:64000 a=rtcp:4017 IN IP4 192.168.1.116 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1567952351 cname:73284f1717045a81 a=ice-ufrag:37e51dc0 a=ice-pwd:49f7442b5078148140877b44 a=candidate:Hc0a80174 1 UDP 2130706431 192.168.1.116 4032 typ host a=candidate:Hc0a80174 2 UDP 2130706430 192.168.1.116 4017 typ host 2025/04/13 15:58:33.430593 192.168.1.75:5060 -> 2.8x.xx.xxx:59396 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.116:59396;rport=59396;branch=z9hG4bKPj14c6f24b814e4d158014cc80b1fa5f8f;received=2.8x.xx.xxx From: "6001" <sip:6005@192.168.1.75:5060>;tag=0397d2bdc3ef47f5b6e498126a360b0f To: <sip:100@192.168.1.75:5060> Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8964 INVITE Server: kamailio (5.7.4 (x86_64/linux)) Content-Length: 0 2025/04/13 15:58:33.461894 192.168.1.75:5060 -> 192.168.1.190:32210 INVITE sip:100@192.168.1.75 SIP/2.0 Via: SIP/2.0/UDP 62.2x.2xx.xxx:5060;branch=z9hG4bK3362.72f5db87deeeccee7fea924753af3da0.0 Via: SIP/2.0/UDP 192.168.1.116:59396;received=2.8x.xx.xxx;rport=59396;branch=z9hG4bKPj14c6f24b814e4d158014cc80b1fa5f8f Max-Forwards: 69 From: "6001" <sip:6005@192.168.1.75:5060>;tag=0397d2bdc3ef47f5b6e498126a360b0f To: <sip:100@192.168.1.75:5060> Contact: "6001" <sip:6005@2.8x.xx.xxx:59396;ob>;+sip.ice Call-ID: 83a804569c3f489088607c729485cdbd CSeq: 8964 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Authorization: Digest username="6005", realm="asterisk", nonce="1744559913/dab8894e3f31e038d7a969a6349a11c2", uri="sip:100@62.2x.2xx.xxx", response="2ce500d6a1583f05f6344bdf1f712fc7", algorithm=MD5, cnonce="808fc766135d40589dd68032bcaab5c6", opaque="2232aa9c42080f12", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 322 v=0 o=- 3953552313 3953552313 IN IP4 192.168.1.75 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 12156 RTP/AVP 8 0 101 c=IN IP4 192.168.1.75 b=TIAS:64000 a=ssrc:1567952351 cname:73284f1717045a81 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:12157 Please Help me, Thank you.

4 8

TCP timeout
by Pyry Aaltonen 11 Apr '25

11 Apr '25

Hello, I´ve googled around about tcp timeout in kamailio tcp connections when the tcp connection is broken. Found this old question and to me it feels like I’m having that same situation as original question https://www.mail-archive.com/sr-users@lists.kamailio.org/msg15020.html Also found this https://kamailio.org/mailman3/hyperkitty/list/sr-dev@lists.kamailio.org/thr… and it seems to be the same description pretty much that I´m seeing. When the tcp connection (or tls) is interrupted in the network it takes around 15min for kamailio to reset the outgoing tcp connection. I see in the logs that when restarting kamailio process it logs 2025-03-27 12:25:18.697 { "level": "INFO", "module": "core", "file": "core/tcp_main.c", "line": 3282, "function": "tcp_init", "logprefix": "", "message": "Set TCP_USER_TIMEOUT=10000 ms" } So I think the fix from this https://github.com/kamailio/kamailio/commit/d893f3af1444c8c4c5db6cd53fb5777… is applied. This is tested with 5.8.5 and I have tested this by setting up with dispatcher tcp connection to external host Then with iptables drop traffic to that host, waiting kamailio to notice that destination is down, removing the iptables input and it takes around 15min to recover. (also restarting the kamailio helps and resolves the connection) And this is what kamailio prints during the test: Apr 4 08:29:15 kamailio[156947]: { "level": "ERROR", "module": "xlog", "file": "xlog.c", "line": 278, "function": "", "logprefix": "", "message": "Destination down: OPTIONS sip:ext-host;transport=tcp (<null>)" } Apr 4 08:44:43 kamailio[156957]: { "level": "ERROR", "module": "core", "file": "core/tcp_read.c", "line": 267, "function": "tcp_read_data", "logprefix": "", "message": "error reading: Connection timed out (110) ([]:5060 -> []:47492)" } Apr 4 08:44:43 kamailio[156957]: { "level": "ERROR", "module": "core", "file": "core/tcp_read.c", "line": 1524, "function": "tcp_read_req", "logprefix": "", "message": "error reading - c: 0x7f8625dea9b0 r: 0x7f8625deaad8 (-1)" } Apr 4 08:45:05 kamailio[156957]: { "level": "ERROR", "module": "xlog", "file": "xlog.c", "line": 278, "function": "", "logprefix": "Source:[ext-host]:5060, Call-id:4d0e1ccd315c817f-156947@int-host, CSeq:10", "message": "Destination up: OPTIONS sip:ext.host;transport=tcp (<null>)" } Any advice how to lower the timeout to be quicker in such event? -Pyry

5 6

leave
by Mihail D. 11 Apr '25

11 Apr '25

leave

1 0

STUN Server vs. NAT Configuration with Kamailio + RTPengine Setup
by Fernando Lopes 11 Apr '25

11 Apr '25

Previous Thread: https://kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio.org/t… Hello everyone, Should I use a STUN server, or can I just configure NAT properly? Here’s my current setup: arduino Copy Edit Router (NAT device) ↓ Kamailio + RTPengine machine ↓ Virtual machine running Kubernetes ↓ Asterisk container Kamailio and RTPengine are sitting behind the NAT (the router), and they handle SIP and media. Do I need a STUN server in this case, or can I manage everything with NAT configuration alone? And what is the better approach? Any insights or best practices would be appreciated! Thank you.

2 1

IPv6 bind wrong IP. #4199
by vunm.allsign＠gmail.com 11 Apr '25

11 Apr '25

Hi guys, I opened issue on github: https://github.com/kamailio/kamailio/issues/4199 - Description: Kamailio (5.8.x): SIP proxy FreeSWITCH: Media - OS: Debian 12 gateway: aaaa:bbb:cccc:20::1 ipv6: aaaa:bbb:cccc:20::41e - kamailio.cfg: #!define ADDR_IPV4 xxx.xxx.xxx.xxx #!define ADDR_IPV6 [aaaa:bbb:cccc:20::41e] listen=ADDR_IPV4 listen=ADDR_IPV6 - I execute netstat -nlutp: tcp6 0 0 aaaa:bbb:cccc:20:::5060 :::* LISTEN 1691274/kamailio tcp6 0 0 aaaa:bbb:cccc:20:::5061 :::* LISTEN 1691274/kamailio udp6 0 0 aaaa:bbb:cccc:20:::5060 :::* 1691222/kamailio Why I set listen ip is aaaa:bbb:cccc:20::41e but kamailio stripped the last part of the address aaaa:bbb:cccc:20:: I have tried many ways but it does not work.Is there any additional configuration required regarding the Debian network interface? - Here is interfaces(of course i can ping from/to ipv6 address): # Additional interfaces, just in case we're using # multiple networks allow-hotplug eth1 iface eth1 inet6 static address aaaa:bbb:cccc:20::41e netmask 64 gateway aaaa:bbb:cccc:20::1 autoconf 0 Thank you.

2 1

Load multiple geoip2 databases (City + ASN)
by Joel Serrano 10 Apr '25

10 Apr '25

Hi guys, Is there a way to have multiple geoip2 databases loaded into Kamailio and then point the geoip2_match to them? If I understand correctly the docs I can only load one: https://www.kamailio.org/docs/modules/6.0.x/modules/geoip2.html#geoip2.p.pa… Is my assumption correct? Thanks, Joel.

3 5

Kamailio + RTP Engine question
by fernandolopes20003＠gmail.com 08 Apr '25

08 Apr '25

Hi everyone, I'm experiencing some issues with Kamailio and RTP Engine. Here's the setup: A user initiates a call, and the traffic goes through a router. The router forwards the traffic to a machine at 192.168.1.75, which runs Kamailio and RTP Engine. Kamailio uses a dispatcher to forward calls to 192.168.1.190, a Kubernetes node running an Asterisk container. The problem: When I'm inside the network, everything works fine — I can send and receive audio. But when I'm outside the network, the call connects, but there's no audio in either direction (no send or receive). Some Configs: Kamailio.conf: modparam("rtpengine", "rtpengine_sock", "udp:0.0.0.0:2223") listen=udp:192.168.1.75:5060 listen=tcp:192.168.1.75:5060 listen=udp:127.0.0.1:5060 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 60) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_timer_mode", 1) rtpengine.conf: interface = 192.168.1.75 listen-udp = 0.0.0.0:2223 port-min = 10000 port-max = 20000 Firewall: To Action From -- ------ ---- 22 ALLOW Anywhere 12222/udp ALLOW Anywhere 5060/udp ALLOW Anywhere 10000:20000/udp ALLOW Anywhere 22 (v6) ALLOW Anywhere (v6) 12222/udp (v6) ALLOW Anywhere (v6) 5060/udp (v6) ALLOW Anywhere (v6) 10000:20000/udp (v6) ALLOW Anywhere (v6) Also router has all this ports open Thank you.

5 6

Question: Kamailio + Rtpengine + NAT
by Fernando Lopes 08 Apr '25

08 Apr '25

Previous Thread: https://kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio.org/t… I've followed some suggestions and got to check sip packets and got this. Change some configs, but couldn't get any NAT configuration to work, hoping to get a new opinion if it still, maybe it is a problem with NAT or maybe another thing. #-------------------------------------------------------------------------------------------------------# #-------------------------------------------------------------------------------------------------------# Called from "outside" and my microsip doesn't have STUN Server on. 2025/04/08 12:19:06.747644 2.81.7x.1xx:59562 -> 192.168.1.75:5060 INVITE sip:100@192.168.1.75:5060 SIP/2.0 Via: SIP/2.0/UDP 2.81.7x.1xx:59562;rport;branch=z9hG4bKPj19931b6928594c3b90ba976acb74cd02 Max-Forwards: 70 From: "6002" <sip:6002@192.168.1.75:5060>;tag=7d887491eae24520b8b205a7dcdf569d To: <sip:100@192.168.1.75:5060> Contact: <sip:6002@2.81.7x.1xx:59562;ob>;+sip.ice Call-ID: fa30f64e6adb4b7e9719084b9d34b5b6 CSeq: 8949 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Content-Type: application/sdp Content-Length: 536 v=0 o=- 3953107147 3953107147 IN IP4 192.168.1.116 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4030 RTP/AVP 8 0 101 c=IN IP4 192.168.1.116 b=TIAS:64000 a=rtcp:4029 IN IP4 192.168.1.116 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:371808528 cname:7dd13e4d736d4043 a=ice-ufrag:69525f90 a=ice-pwd:16496df15af141bb26e901eb a=candidate:Hc0a80174 1 UDP 2130706431 192.168.1.116 4030 typ host a=candidate:Hc0a80174 2 UDP 2130706430 192.168.1.116 4029 typ host #-------------------------------------------------------------------------------------------------------# #-------------------------------------------------------------------------------------------------------# With STUN Server On -- Also get Register but still can't get any audio 2025/04/08 12:18:26.032585 2.81.7x.1xx:57124 -> 192.168.1.75:5060 INVITE sip:100@192.168.1.75:5060 SIP/2.0 Via: SIP/2.0/UDP 2.81.7x.1xx:57124;rport;branch=z9hG4bKPjaf3ec8e03d52425eb67708c7e61457b1 Max-Forwards: 70 From: "6002" <sip:6002@192.168.1.75:5060>;tag=10bf4ad8533a4a45b478c900cb3242b1 To: <sip:100@192.168.1.75:5060> Contact: "6002" <sip:6002@2.81.7x.1xx:57124;ob>;+sip.ice Call-ID: 1e3dea0a3d2f4954b4232c624d37c9d8 CSeq: 7501 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.21.5 Content-Type: application/sdp Content-Length: 726 v=0 o=- 3953107106 3953107106 IN IP4 2.81.7x.1xx s=pjmedia b=AS:84 t=0 0 a=X-nat:1 m=audio 4033 RTP/AVP 8 0 101 c=IN IP4 2.81.7x.1xx b=TIAS:64000 a=rtcp:4035 IN IP4 2.81.7x.1xx a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:803428650 cname:24b959022cf70edd a=ice-ufrag:072851d1 a=ice-pwd:10d96c6c6ea14c665c5e6d4e a=candidate:Sc0a80174 1 UDP 1862270975 2.81.7x.1xx 4033 typ srflx raddr 192.168.1.116 rport 4033 a=candidate:Hc0a80174 1 UDP 1694498815 192.168.1.116 4033 typ host a=candidate:Sc0a80174 2 UDP 1862270974 2.81.7x.1xx 4035 typ srflx raddr 192.168.1.116 rport 4035 a=candidate:Hc0a80174 2 UDP 1694498814 192.168.1.116 4035 typ host Thank you Very much.

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users