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37474 discussions

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

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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

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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

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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

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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!

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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

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

21 May '25

Hello, I am considering to release Kamailio v5.7.7 (out of branch 5.7) on Wednesday, May 21, 2025, as the last one from the 5.7.x series, to mark the end of support for it. Currently 5.8.x and 6.0.x are the latest two maintained branches. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

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Re: stir/shaken integration with clearip redirect method
by JR Richardson 20 May '25

20 May '25

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

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

19 May '25

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

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Server closing TLS connection
by Ihor Olkhovskyi 16 May '25

16 May '25

Hello, I'm having a strange issue, when Kamailio(?) is closing TLS connection just in the middle of session. Usually I see a sort of timeout ~ 600 secs. But it does not ring a bell for me. With strace I've found, that it's actually not Kamailio, but underlying SSL library is doing this, according to line I've got from strace [2025-05-16 10:00:30] PID=1337952 FD=0 SYSCALL=close -> [No socket info] Raw: sendto(3, "<183>May 16 10:00:30 /usr/local/sbin/kamailio[1337952]: DEBUG: tls [tls_server.c:712]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7f01a1ab1870\n", 150, MSG_NOSIGNAL, NULL, 0) = 150 As I don't see any explicit calls of tls_h_tcpconn_close_f in the source code, I assume it's a callback from the external OpenSSL(?) module. What would be the best way to debug why OpenSSL(?) is terminating connection, cause on Wireshark I just do see usual TCP connection with keepalives [image: image.png] kamailio -v version: kamailio 5.8.3 (x86_64/linux) 6f8a04 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. openssl --version OpenSSL 3.2.2 4 Jun 2024 (Library: OpenSSL 3.2.2 4 Jun 2024) enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=30 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=5 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_script_mode=1 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 Many thanks in advance! -- Best regards, Ihor (Igor)

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