- sr-users - kamailio.org

kamailio call redirect
by fabienbrou＠alcall.net 26 Mar '25

26 Mar '25

Hello everyone. I'm trying to redirect a call directly through kamailio by modifying the $ru variable. Just wondering if this is the right method or if there is a better way to do it. thanks

4 13

is possible empty actions?
by Sergey Safarov 26 Mar '25

26 Mar '25

I want to use contributions like route[PROXY_MEDIA] { if (is_method("BYE|CANCEL")) { # passing message for RTPengine processing } else if (!sdp_content()) { return; } # more instructions here ... rtpengine_manage(); } Now Kamailio cannot start because first if block has empty actions. How does nothing in the actions? I want to avoid not needed log messages and do not use constructions like if (!sdp_content() and ! is_method("BYE|CANCEL")) { return; }

3 2

Seeking advices on redesigning our VoIP system for scalability and manageability
by Francesco 26 Mar '25

26 Mar '25

Dear all, I hope this email finds you well. I am reaching out to gather insights and suggestions regarding a potential redesign of our VoIP system to improve scalability, ease of management, and upgradeability. *Current Setup:* - Our system is *not multi-tenant* and is based on *Kamailio and FreeSWITCH* in a distributed architecture (as example of distributed architecture, see: https://ufdcimages.uflib.ufl.edu/UF/E0/04/92/89/00001/THOMPSON_C.pdf). - Everything runs inside *LXC containers* across multiple nodes. - One of the challenges we face is *OS upgrades*, which are cumbersome and require significant manual intervention. - At the moment the code for voip is done with ansible Given these challenges, I am considering whether a *microservices-based architecture* could be a viable option. However, I have some concerns regarding: - *SIP and RTP handling with NATed containers* (e.g., Kubernetes, Docker Swarm) and potential issues with media traversal. - *Performance impact* when shifting real-time traffic handling to containerized environments. - *Upgrade and deployment strategies*—how to achieve smoother, rolling upgrades without service interruptions. - *Alternative approaches* that might be more effective, such as a hybrid model combining LXC with containerized microservices for specific functions. If anyone has experience transitioning a VoIP system to microservices or has insights on best practices for *scalable, maintainable VoIP architectures*, I would greatly appreciate your thoughts. What worked for you? What pitfalls should we be aware of?Do you have any documentation that I can check? Looking forward to learning from your experiences. Thank you. .: Francesco Colista .: Francesco Colista

3 2

TOPOS Issue with Response Handling on two different Interfaces
by Vinayak Makwana 25 Mar '25

25 Mar '25

Hello all, We are facing an issue while configuring TOPOS in Kamailio P-CSCF (running under Docker). Our objective is to strip Via, Record-Route, and Route headers for both INVITE and 180 Ringing responses. However, we observe unexpected behavior when handling the 180 Ringing/200 OK responses. Call Flow & Issue Description: Note: Public IP and Private IP mapping using the below configuration. listen=udp:Local_IP:5060 name "internal" listen=udp:Local_IP:6070 advertise Public_IP:6070 name "external" Call Flow (INVITE): A-Party --> Kamailio P-CSCF (Public IP) --> Kamailio P-CSCF (Private IP) --> Kamailio S-CSCF (Private IP) --> Kamailio P-CSCF (Private IP) --> Kamailio P-CSCF (Public IP) --> B-Party For an incoming *INVITE* (A-Party → B-Party), TOPOS successfully strips the headers when the message flows through the P-CSCF, ensuring proper privacy of routing information. *Issue* : When B-Party sends a 180 Ringing, we observe that Kamailio P-CSCF (Private IP) adds a Record-Route header before forwarding the message to A-Party. Call Flow (180 Ringing): B-Party --> Kamailio P-CSCF (Public IP) --> Kamailio P-CSCF (Private IP) --> Kamailio S-CSCF (Private IP) We need to ensure that the Record-Route and Via headers are stripped properly in both directions. Please guide us how to achieve and what we are doing wrong. TOPOS module is configured as follows: modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379") modparam("topos_redis", "serverid", "srv1") modparam("topos", "storage", "redis") modparam("topos", "contact_mode", 1) modparam("topos", "rr_update", 1) RR module is configured as follows: modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) configured the below event_route: event_route[topos:msg-outgoing] { xlog("L_INFO","[topos:msg-outgoing]"); if(is_request() and $sndto(ip)=="SCSCF_PRIVATE_IP") { xlog("L_INFO","[topos:msg-outgoing] TOPOS \n"); drop; } } Thanks, Vinayak Makwana -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

1 1

Running Kamailio in kubernetes
by sagar agarwal 25 Mar '25

25 Mar '25

Hi team, I am looking for some guidance to run Kamailio as part of our Kubernetes cluster. I don't see any official or well maintained helm chart out there to install Kamailio on K8s. If there is a recommendation from your side on a helm chart please let me know. Also would be interesting to know if you foresee any issues with running Kamailio on k8s or do you feel its better to stick with ec2 based installations. Regards Sagar Agarwal

4 3

Problem with ipsec setting of Via for outgoing message
by Minello, Loredana 24 Mar '25

24 Mar '25

Hello, I'm using Kamailio ver. 5.8.4. I configured the ims_ipsec_pcscf with modparam("ims_ipsec_pcscf", "ipsec_max_connections", 1) and call ipsec_forward with flag = IPSEC_FORWARD_FLAGS where : # - IPSEC_SEND_FORCE_SOCKET 1 # - IPSEC_FORWARD_USEVIA 256 #!define IPSEC_FORWARD_FLAGS 257 For the UE REGISTRATION I have the following flow UE PCSCF_GM_IMS REGISTER (32133) ------------------> (5060) 100 Trying (32133) <------------------ (5060) 401 Unauthorized (32133) <------------------ (5060) REGISTER (TCP) (PORT_UC) ------------------> (PORT_PS) 100 Trying (TCP) (PORT_UC) <------------------ (PORT_PS) 200 OK (TCP) (PORT_UC) <------------------ (PORT_PS) ------------------------------------------------------------------ SUBSCRIBE (UDP) (PORT_UC) ------------------> (PORT_PS) 200 Subscription (UDP) (PORT_US) <------------------ (PORT_PC) NOTIFY (UDP) (PORT_US) <------------------ (PORT_PC) 200 OK (UDP) (PORT_UC) ------------------> (PORT_PC) NOTIFY (UDP) (PORT_US) <------------------ (PORT_PC) 200 OK (UDP) (PORT_UC) ------------------> (PORT_PC) NOTIFY (UDP) (PORT_US) <------------------ (PORT_PC) 200 OK (UDP) (PORT_UC) ------------------> (PORT_PC) The SUBSCRIBE behavior is correct : PCSCF receives message on PORT_PS and sends on PORT_PC The NOTIFY behavior is wrong: PCSCF send from PORT_PC (correct), but UE respond towards PORT_PC, defined in the Via of the NOTIFY. I know I can use the set_advertised_port() function, and in this case can work, but the function is taking a constant as a parameter, not a variable. This is causing a problem when ipsec_max_connections > 1. In that case it won't be easy to pass the correct port. I was wondering if the ipsec module should properly set the VIA port to the correct PORT_PS, rather than leaving the setting of the correct port by the kamailio.cfg. Do you have any hints on that? Thanks

1 0

Re: send_fd(): sendmsg failed sending 55 on 52: Too many references: cannot splice (109)
by Mathias Schneuwly 24 Mar '25

24 Mar '25

Thank you, I've already changed the value, but that didn't solve my issue with the blocking Kamailio. I did some tests with an older Yocto image which contains Kamailio 5.5.4 and OpenSSL 1.1.1w. With this combination I was not able to reproduce the deadlock. Am Fr., 21. März 2025 um 16:49 Uhr schrieb Harald Kapper <hk(a)kapper.net>: > Hi > > fyi – in case you need this: > > systemctl edit kamailio.service > > > > add: > > ### Editing /etc/systemd/system/kamailio.service.d/override.conf > > ### Anything between here and the comment below will become the contents > of the drop-in file > > > > [Service] > > LimitNOFILE=40960 > > > > ### Edits below this comment will be discarded > > > > (or whatever number you’d like) > > saving saves the override for systemd into (like) > /etc/systemd/system/kamailio.service.d/override.conf > > > > Afterwards you have to restart the service (systemctl restart > kamailio.service) > > > > And you should be fine. > > > > Regards, hk > > > > *Von:* Henning Westerholt via sr-users <sr-users(a)lists.kamailio.org> > *Gesendet:* Donnerstag, 20. März 2025 14:25 > *An:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > *Cc:* Mathias Schneuwly <mathias(a)schneuwlys.ch>; Henning Westerholt < > hw(a)gilawa.com> > *Betreff:* [SR-Users] Re: send_fd(): sendmsg failed sending 55 on 52: Too > many references: cannot splice (109) > > > > Hello Mathias, > > > > yes, you can set the open files e.g. in the systemd unit file for Kamailio. > > > > Cheers, > > > > Henning > > > > *From:* Mathias Schneuwly via sr-users <sr-users(a)lists.kamailio.org> > *Sent:* Donnerstag, 20. März 2025 13:43 > *To:* sr-users(a)lists.kamailio.org > *Cc:* Mathias Schneuwly <mathias(a)schneuwlys.ch> > *Subject:* [SR-Users] Re: send_fd(): sendmsg failed sending 55 on 52: Too > many references: cannot splice (109) > > > > > I'll also give a try with 6.0.1 later. > > Meanwhile, I've tested 5.8.6 and 6.0.1 and both show the same issue. > > > > Am Do., 20. März 2025 um 10:21 Uhr schrieb Mathias Schneuwly < > mathias(a)schneuwlys.ch>: > > Hi Henning > > > > > Maybe also check the actual limits by “cat /proc/$kamailio-pid/limits” > Not 100% if its related, but if you are looking to scale this would be for > sure required. > > The limit was > > Max open files 1024 524288 files > > and I changed it now to > > Max open files 65536 65536 files > > > > Are there some recommendations for the limit? Would it make sense to > change that in the provided systemd file if you say that 1024 is not that > much? > > > > But anyway, it fails with 1024 and with 65536. > > > > I've also updated to 5.8.5 and it shows the same issue. I'll also give a > try with 6.0.1 later. > > > > > About the tls_thread_mode – you should use mode 2 if you are running > with openssl 3.x, otherwise you probably will observe random crashes. > > We are running OpenSSL 3.2.3 and tls_thread_mode is set to 2. But it still > blocks. > > > > Best regards > > Mathias > > > > Am Do., 20. März 2025 um 08:42 Uhr schrieb Henning Westerholt < > hw(a)gilawa.com>: > > Hello Mathias, > > > > 1024 open files are not that much, you should increase it for the kamailio > user. Maybe also check the actual limits by “cat > /proc/$kamailio-pid/limits” > > Not 100% if its related, but if you are looking to scale this would be for > sure required. > > > > About the tls_thread_mode – you should use mode 2 if you are running with > openssl 3.x, otherwise you probably will observe random crashes. > > > > If you are seeing the mentioned problem with 15 phones, this is clearly a > problem and should be further investigated. > > > > Cheers, > > > > Henning > > > > > > *From:* Mathias Schneuwly via sr-users <sr-users(a)lists.kamailio.org> > *Sent:* Donnerstag, 20. März 2025 08:07 > *To:* sr-users(a)lists.kamailio.org > *Cc:* Mathias Schneuwly <mathias(a)schneuwlys.ch> > *Subject:* [SR-Users] Re: send_fd(): sendmsg failed sending 55 on 52: Too > many references: cannot splice (109) > > > > Hi > > > > Thanks for the reply. We are using the following limits and I guess they > are the default values > > real-time non-blocking time (microseconds, -R) unlimited > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) 15666 > max locked memory (kbytes, -l) 8192 > max memory size (kbytes, -m) unlimited > open files (-n) 1024 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) 819200 > real-time priority (-r) 0 > stack size (kbytes, -s) 8192 > cpu time (seconds, -t) unlimited > max user processes (-u) 15666 > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > > But to me, it looks more like a deadlock somewhere around SSL_accept. > According to the stack trace generated with "kamctl trap", all 8 TCP > receiver processes are waiting in "#1 0x00007fa14ca04830 in > pthread_rwlock_wrlock () from /usr/lib/libc.so.6". I guess the message "Too > many references: cannot splice (109)" is just the consequence as the main > process is no longer able to dispatch jobs to the child workers. > > > > Meanwhile, I'm able to reproduce the issue. It requires a bit of brute > force, but it deadlocks reliably after a few seconds. > > > > I'm flooding Kamailio with register messages using sipexer. I'm running > the following command 3 times in parallel and always after a few seconds, > Kamailio is completely blocked. > > watch -n 0.1 "./sipexer -register -vl 3 -co -com -ex 3600 -fu "2" -au "2" > -ap "test" -tk <path to key>.key -tc <path to cert>.cer -ti -sd -ru > sip:<domain> -cu sip:2@<domain> tls:<fqdn>:5061;sleep 0.1;./sipexer > -register -vl 3 -co -com -ex 0 -fu "2" -au "2" -ap "test" -tk <path to > key>.key -tc <path to cert>.cer -ti -sd -ru sip:<domain> -cu sip:2@<domain> > tls:<fqdn>:5061" > > > > I played a bit with the parameters "modparam("tls", "init_mode", 1)" and > "tls_threads_mode = 2", but it didn't seem to change anything. > > > > Please have in mind that we usually do not stress Kamailio that much. We > have seen the issue a couple of times with around 5 to 15 SIP phones > registered. But in this usual setup, the problem does not occur reliably. > > > > Best regards > > Mathias > > > > Am Di., 18. März 2025 um 21:53 Uhr schrieb Henning Westerholt < > hw(a)gilawa.com>: > > Hello, > > > > check the ulimit settings for the kamailio user and general resource usage > on that system. In development branch there is also a feature present to > print the actual ulimits on startup. > > > > Cheers, > > > > Henning > > > > *From:* Mathias Schneuwly via sr-users <sr-users(a)lists.kamailio.org> > *Sent:* Montag, 17. März 2025 13:41 > *To:* sr-users(a)lists.kamailio.org > *Cc:* Mathias Schneuwly <mathias(a)schneuwlys.ch> > *Subject:* [SR-Users] send_fd(): sendmsg failed sending 55 on 52: Too > many references: cannot splice (109) > > > > Hi > > > > We are using kamailio 5.8.4. > > > > Recently, we noticed, that after a few days, we see the following messages > from Kamailio `2025 Mar 17 10:22:11 ttel CRITICAL: <core> > [core/pass_fd.c:193]: send_fd(): sendmsg failed sending 55 on 52: Too many > references: cannot splice (109)`. > > > > When this happens, devices are no longer able to register. At the moment > we have no idea what might be wrong. > > > > Unfortunately, the internet does not reveal much information regarding > this issue. In one issue related to Kamailio ( > https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…), > I found the command `kamctl trap` and when I execute that, I see one issue, > which might be related, but I'm just guessing... > > > > ``` > > ---start 1782236 ----------------------------------------------------- > [Thread debugging using libthread_db enabled] > Using host libthread_db library "/usr/lib/libthread_db.so.1". > 0x00007fa14c9fb1c6 in ?? () from /usr/lib/libc.so.6 > $1 = 26 > $2 = {pid = 1782236, unix_sock = 36, idx = 0, status = 1, rank = 17, desc > = "tcp receiver (generic) child=0", '\000' <repeats 97 times>} > #0 0x00007fa14c9fb1c6 in ?? () from /usr/lib/libc.so.6 > No symbol table info available. > #1 0x00007fa14ca04830 in pthread_rwlock_wrlock () from /usr/lib/libc.so.6 > No symbol table info available. > #2 0x00007fa14ac20b49 in CRYPTO_THREAD_write_lock () from > /usr/lib/libcrypto.so.3 > No symbol table info available. > #3 0x00007fa14adbd5da in ?? () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #4 0x00007fa14adbd9ca in ?? () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #5 0x00007fa14adbc907 in ?? () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #6 0x00007fa14adbd26c in ?? () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #7 0x00007fa14adbd68e in ?? () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #8 0x00007fa14abe75c6 in ?? () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #9 0x00007fa14abe840b in EVP_RAND_generate () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #10 0x00007fa14acf9282 in RAND_bytes_ex () from /usr/lib/libcrypto.so.3 > No symbol table info available. > #11 0x00007fa14b06cce8 in ?? () from /usr/lib/libssl.so.3 > No symbol table info available. > #12 0x00007fa14b05bdf7 in ?? () from /usr/lib/libssl.so.3 > No symbol table info available. > #13 0x00007fa14b0ef036 in tls_accept (c=c@entry=0x7fa1475c0dc0, > error=error@entry=0x7ffebd9443b8) at tls_server.c:457 > ret = <optimized out> > ssl = warning: could not convert 'ssl_st' from the host encoding > (ANSI_X3.4-1968) to UTF-32. > This normally should not happen, please file a bug report. > 0x7fa147668bd0 > cert = <optimized out> > tls_c = 0x7fa14768f490 > tls_log = <optimized out> > __func__ = "tls_accept" > #14 0x00007fa14b0f7540 in tls_h_read_f (c=0x7fa1475c0dc0, > flags=0x7ffebd9647d0) at tls_server.c:1153 > r = 0x7fa1475c0ee8 > bytes_free = 16383 > bytes_read = <optimized out> > read_size = <optimized out> > ssl_error = 0 > ssl_read = 0 > ssl = 0x7fa147668bd0 > rd_buf = > "\026\003\003\004\016\v\000\004\n\000\004\a\000\004\0040\202\004\0000\202\002\350\240\003\002\001\002\002\002\026\0270\r\006\t*\206H\206\367\r\001\001\v\005\0000b1\v0\t\006\003U\004\006\023\002CH1\r0\v\006\003U\004\b\f\004Bern1\r0\v\006\003U\004\a\f\004Bern1\r0\v\006\003U\004\n\f\004RUAG1\0200\016\006\003U\004\v\f\aDefence1\0240\022\006\003U\004\003\f\versaimfs.ch0\036\027\r230302163519Z\027\r291230163519Z0`1\v0\t\006\003U\004\006\023\002CH1\r0\v\006\003"... > wr_buf = > "\026\003\003\000A\002\000\000=\003\003\371\204\326\361\262-9\300\344gV\2316\373\235D\314\225u8\v}\215\254DOWNGRD\001\000\3000\000\000\025\377\001\000\001\000\000\v\000\004\003\000\001\002\000#\000\000\000\027\000\000\026\003\003\a\267\v\000\a\263\000\a\260\000\004\0040\202\004\0000\202\002\350\240\003\002\001\002\002\002\r>0\r\006\t*\206H\206\367\r\001\001\v\005\0000b1\v0\t\006\003U\004\006\023\002CH1\r0\v\006\003U\004\b\f\004Bern1\r0\v\006\003U\004\a\f\004Bern1\r0\v\006\003U\004\n\f\004RUAG1\0200\016\006\003U\004\v\f\aDefence1\0240\022\006"... > rd = {buf = 0x7ffebd944480 "\026\003\003\004\016\v", pos = 1405, > used = 1405, size = 65536} > wr = {buf = 0x7ffebd954480 "\026\003\003", pos = 0, used = 0, size > = 65536} > tls_c = 0x7fa14768f490 > enc_rd_buf = <optimized out> > n = 0 > flush_flags = 0 > err_src = <optimized out> > ip_buf = '\000' <repeats 63 times> > x = <optimized out> > tls_dbg = <optimized out> > __func__ = "tls_h_read_f" > redo_read = <optimized out> > continue_ssl_read = <optimized out> > __llevel = <optimized out> > __kld = <optimized out> > __llevel = <optimized out> > __kld = <optimized out> > #15 0x000055d2519871b4 in tcp_read_headers (c=c@entry=0x7fa1475c0dc0, > read_flags=read_flags@entry=0x7ffebd9647d0) at core/tcp_read.c:445 > bytes = <optimized out> > remaining = <optimized out> > p = <optimized out> > r = 0x7fa1475c0ee8 > mc = <optimized out> > body_len = <optimized out> > tvnow = {tv_sec = 1742100857, tv_usec = 944923} > tvdiff = <optimized out> > mfline = <optimized out> > mtransid = <optimized out> > __func__ = "tcp_read_headers" > #16 0x000055d251989cef in tcp_read_req (con=0x7fa1475c0dc0, > bytes_read=bytes_read@entry=0x7ffebd9647c8, read_flags=read_flags@entry=0x7ffebd9647d0) > at core/tcp_read.c:1509 > bytes = <optimized out> > total_bytes = 0 > resp = 1 > size = <optimized out> > req = 0x7fa1475c0ee8 > dst = {send_sock = 0x55d251a91020, to = {s = {sa_family = 52024, > sa_data = "\257Q\322U\000\000\002\000\000\000\376\177\000"}, sin = > {sin_family = 52024, sin_port = 20911, sin_addr = {s_addr = 21970}, > sin_zero = "\002\000\000\000\376\177\000"}, sin6 = {sin6_family = 52024, > sin6_port = 20911, sin6_flowinfo = 21970, sin6_addr = {__in6_u = > {__u6_addr8 = > "\002\000\000\000\376\177\000\000\030\000\000\000\000\000\000", __u6_addr16 > = {2, 0, 32766, 0, 24, 0, 0, 0}, __u6_addr32 = {2, 32766, 24, 0}}}, > sin6_scope_id = 0}, sas = {ss_family = 52024, __ss_padding = > "\257Q\322U\000\000\002\000\000\000\376\177\000\000\030", '\000' <repeats > 23 times>, > "\001\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000\262+\226Q\001 > \000\000\200,5L\241\177\000\000 \020\251Q\322U\000\000 > \020\251Q\322U\000\000\200\303\257Q\322U\000\00009SG\241\177\000\000\330G\226\275\376\177\000\000\b\000\000\000\000\000\000", > __ss_align = 20}}, id = 1, send_flags = {f = 1, blst_imask = 0}, proto = 14 > '\016', proto_pad0 = 0 '\000', proto_pad1 = 0} > c = <optimized out> > ret = <optimized out> > again = <optimized out> > __func__ = "tcp_read_req" > #17 0x000055d25198ebec in handle_io (fm=fm@entry=0x7fa14c352c80, > events=events@entry=1, idx=idx@entry=-1) at core/tcp_read.c:1906 > ret = 8 > n = 0 > read_flags = RD_CONN_SHORT_READ > con = 0x7fa1475c0dc0 > s = 14 > resp = <optimized out> > t = <optimized out> > ee = 0x0 > __func__ = "handle_io" > error = <optimized out> > #18 0x000055d25199469b in io_wait_loop_epoll (repeat=repeat@entry=0, t=2, > h=0x55d251c49d60 <io_w>) at core/io_wait.h:1075 > n = 1 > r = 0 > fm = 0x7fa14c352c80 > revents = 1 > __func__ = "io_wait_loop_epoll" > #19 0x000055d251994e87 in tcp_receive_loop (unix_sock=<optimized out>) at > core/tcp_read.c:2026 > __func__ = "tcp_receive_loop" > #20 0x000055d25197e72c in tcp_init_children (woneinit=woneinit@entry=0x7ffebd964d1c) > at core/tcp_main.c:5391 > r = <optimized out> > i = <optimized out> > reader_fd_1 = 39 > pid = <optimized out> > si_desc = "tcp receiver > (generic)\000\000`M\226\275\376\177\000\000\001\000\000\000\000\000\000\000\210\035\223Q\322U\000\000 > \020\251Q\322U\000\000\220\315\257Q\322U\000\000\220\242/L\241\177\000\000\025\000\000\000\000\000\000\000\004\000\000\000\000\000\000\000%\020\251Q\322U\000\000HU\257Q\322U\000\000e\005\000\000\000\000\000\000 > \020\251Q\322U\000\000\3750\235Q\322U\000" > si = <optimized out> > __func__ = "tcp_init_children" > error = <optimized out> > #21 0x000055d2517c6c7e in main_loop () at main.c:1950 > i = <optimized out> > pid = <optimized out> > si = <optimized out> > si_desc = "udp receiver child=7 sock=10.0.90.1:5060\000\000\000\000\322U\000\000?\342\017K\241\177\000\000\022\000\000\000\377\377\377\377\200\006\265L\241\177\000\000\253m\237L\241\177\000\000\361|\003\000\000\000\000\000\t\000\000\000\000\000\000\000Sat > Mar 15 05:25:32 2025\n\000\264L\241\177\000" > nrprocs = <optimized out> > woneinit = 1 > __func__ = "main_loop" > error = <optimized out> > #22 0x000055d2517bc151 in main (argc=<optimized out>, argv=<optimized > out>) at main.c:3256 > cfg_stream = <optimized out> > c = <optimized out> > r = <optimized out> > tmp = 0x7ffebd966e7f "" > tmp_len = 32673 > port = 5060 > proto = 0 > aproto = 0 > ahost = 0x0 > aport = 0 > options = 0x55d251aa74b8 > ":f:cm:M:dVIhEeb:B:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" > ret = -1 > seed = 3209119183 > rfd = <optimized out> > debug_save = <optimized out> > debug_flag = <optimized out> > dont_fork_cnt = 0 > n_lst = <optimized out> > p = <optimized out> > st = {st_dev = 24, st_ino = 46237, st_nlink = 2, st_mode = 16888, > st_uid = 899, st_gid = 899, __pad0 = 0, st_rdev = 0, st_size = 40, > st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1742012732, tv_nsec = > 217328715}, st_mtim = {tv_sec = 1742012732, tv_nsec = 217328715}, st_ctim = > {tv_sec = 1742012732, tv_nsec = 217328715}, __glibc_reserved = {0, 0, 0}} > l1 = <optimized out> > tbuf = '\000' <repeats 32 times>, > "\030\202\237\275\376\177\000\000@\002\000\000@\003\000\000\001", '\000' > <repeats 23 times>, "@\003\000\000@\003\000\000@\003\000\000@\003\000\000@ > \003\000\000@\003\000\000@\003\000\000@\003\000\000@\003\000\000@ > \003\000\000@\003\000\000@\003\000\000@\003\000\000@\003\000\000@ > \003\000\000@\003\000\000@\003\000\000@\003\000\000@\003\000\000@ > \003\000\000@\003\000\000@\003", '\000' <repeats 11 times>, "\001", > '\000' <repeats 118 times>... > option_index = 12 > long_options = {{name = 0x55d251a9230b "help", has_arg = 0, flag = > 0x0, val = 104}, {name = 0x55d251a937a4 "version", has_arg = 0, flag = 0x0, > val = 118}, {name = 0x55d251a9826a "alias", has_arg = 1, flag = 0x0, val = > 1024}, {name = 0x55d251a92310 "subst", has_arg = 1, flag = 0x0, val = > 1025}, {name = 0x55d251a92316 "substdef", has_arg = 1, flag = 0x0, val = > 1026}, {name = 0x55d251a9231f "substdefs", has_arg = 1, flag = 0x0, val = > 1027}, {name = 0x55d251a92329 "server-id", has_arg = 1, flag = 0x0, val = > 1028}, {name = 0x55d251a92333 "loadmodule", has_arg = 1, flag = 0x0, val = > 1029}, {name = 0x55d251a9233e "modparam", has_arg = 1, flag = 0x0, val = > 1030}, {name = 0x55d251a92347 "log-engine", has_arg = 1, flag = 0x0, val = > 1031}, {name = 0x55d251a938c1 "debug", has_arg = 1, flag = 0x0, val = > 1032}, {name = 0x55d251a92352 "cfg-print", has_arg = 0, flag = 0x0, val = > 1033}, {name = 0x55d251a9235c "atexit", has_arg = 1, flag = 0x0, val = > 1034}, {name = 0x55d251a92363 "all-errors", has_arg = 0, flag = 0x0, val = > 1035}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} > __func__ = "main" > [Inferior 1 (process 1782236) detached] > ---end 1782236 ------------------------------------------------------- > > ``` > > > > Does anybody have an idea what this could be? I never saw that with > previous versions, and now we can see it on several instances, but we are > yet unable to reproduce. > > > > Not sure if this might be related, but we also see many messages like `Mar > 17 13:39:03 ttel /usr/sbin/kamailio[2882127]: CRITICAL: <core> > [core/tcp_main.c:5558]: tcp_timer_check_connections(): message processing > timeout on connection id: 402 (state: 0) - closing` > > > > Best regards > > Mathias > >

1 0

Dispatcher module resets URI parameters
by Marat Gareev 22 Mar '25

22 Mar '25

Hello, I encountered a problem that ds_select_domain function does not save URI parameters. I found workaround: if (!ds_select_domain( "1", 4, 3 )) { send_reply("502", "Bad Gateway"); exit; } if (!strempty($(ou{uri.params}))) { $ru = $ru+";"+$(ou{uri.params}); } But this can lead to duplicates if the URI in the dispatcher list contains the same parameter. Is there any other reliable way to keep original URI parameters while using the dispatcher? Thanks

2 1

send_fd(): sendmsg failed sending 55 on 52: Too many references: cannot splice (109)
by Mathias Schneuwly 20 Mar '25

20 Mar '25

Hi We are using kamailio 5.8.4. Recently, we noticed, that after a few days, we see the following messages from Kamailio `2025 Mar 17 10:22:11 ttel CRITICAL: <core> [core/pass_fd.c:193]: send_fd(): sendmsg failed sending 55 on 52: Too many references: cannot splice (109)`. When this happens, devices are no longer able to register. At the moment we have no idea what might be wrong. Unfortunately, the internet does not reveal much information regarding this issue. In one issue related to Kamailio ( https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…), I found the command `kamctl trap` and when I execute that, I see one issue, which might be related, but I'm just guessing... ``` ---start 1782236 ----------------------------------------------------- [Thread debugging using libthread_db enabled] Using host libthread_db library "/usr/lib/libthread_db.so.1". 0x00007fa14c9fb1c6 in ?? () from /usr/lib/libc.so.6 $1 = 26 $2 = {pid = 1782236, unix_sock = 36, idx = 0, status = 1, rank = 17, desc = "tcp receiver (generic) child=0", '\000' <repeats 97 times>} #0 0x00007fa14c9fb1c6 in ?? () from /usr/lib/libc.so.6 No symbol table info available. #1 0x00007fa14ca04830 in pthread_rwlock_wrlock () from /usr/lib/libc.so.6 No symbol table info available. #2 0x00007fa14ac20b49 in CRYPTO_THREAD_write_lock () from /usr/lib/libcrypto.so.3 No symbol table info available. #3 0x00007fa14adbd5da in ?? () from /usr/lib/libcrypto.so.3 No symbol table info available. #4 0x00007fa14adbd9ca in ?? () from /usr/lib/libcrypto.so.3 No symbol table info available. #5 0x00007fa14adbc907 in ?? () from /usr/lib/libcrypto.so.3 No symbol table info available. #6 0x00007fa14adbd26c in ?? () from /usr/lib/libcrypto.so.3 No symbol table info available. #7 0x00007fa14adbd68e in ?? () from /usr/lib/libcrypto.so.3 No symbol table info available. #8 0x00007fa14abe75c6 in ?? () from /usr/lib/libcrypto.so.3 No symbol table info available. #9 0x00007fa14abe840b in EVP_RAND_generate () from /usr/lib/libcrypto.so.3 No symbol table info available. #10 0x00007fa14acf9282 in RAND_bytes_ex () from /usr/lib/libcrypto.so.3 No symbol table info available. #11 0x00007fa14b06cce8 in ?? () from /usr/lib/libssl.so.3 No symbol table info available. #12 0x00007fa14b05bdf7 in ?? () from /usr/lib/libssl.so.3 No symbol table info available. #13 0x00007fa14b0ef036 in tls_accept (c=c@entry=0x7fa1475c0dc0, error=error@entry=0x7ffebd9443b8) at tls_server.c:457 ret = <optimized out> ssl = warning: could not convert 'ssl_st' from the host encoding (ANSI_X3.4-1968) to UTF-32. This normally should not happen, please file a bug report. 0x7fa147668bd0 cert = <optimized out> tls_c = 0x7fa14768f490 tls_log = <optimized out> __func__ = "tls_accept" #14 0x00007fa14b0f7540 in tls_h_read_f (c=0x7fa1475c0dc0, flags=0x7ffebd9647d0) at tls_server.c:1153 r = 0x7fa1475c0ee8 bytes_free = 16383 bytes_read = <optimized out> read_size = <optimized out> ssl_error = 0 ssl_read = 0 ssl = 0x7fa147668bd0 rd_buf = "\026\003\003\004\016\v\000\004\n\000\004\a\000\004\0040\202\004\0000\202\002\350\240\003\002\001\002\002\002\026\0270\r\006\t*\206H\206\367\r\001\001\v\005\0000b1\v0\t\006\003U\004\006\023\002CH1\r0\v\006\003U\004\b\f\004Bern1\r0\v\006\003U\004\a\f\004Bern1\r0\v\006\003U\004\n\f\004RUAG1\0200\016\006\003U\004\v\f\aDefence1\0240\022\006\003U\004\003\f\versaimfs.ch0\036\027\r230302163519Z\027\r291230163519Z0`1\v0\t\006\003U\004\006\023\002CH1\r0\v\006\003"... wr_buf = "\026\003\003\000A\002\000\000=\003\003\371\204\326\361\262-9\300\344gV\2316\373\235D\314\225u8\v}\215\254DOWNGRD\001\000\3000\000\000\025\377\001\000\001\000\000\v\000\004\003\000\001\002\000#\000\000\000\027\000\000\026\003\003\a\267\v\000\a\263\000\a\260\000\004\0040\202\004\0000\202\002\350\240\003\002\001\002\002\002\r>0\r\006\t*\206H\206\367\r\001\001\v\005\0000b1\v0\t\006\003U\004\006\023\002CH1\r0\v\006\003U\004\b\f\004Bern1\r0\v\006\003U\004\a\f\004Bern1\r0\v\006\003U\004\n\f\004RUAG1\0200\016\006\003U\004\v\f\aDefence1\0240\022\006"... rd = {buf = 0x7ffebd944480 "\026\003\003\004\016\v", pos = 1405, used = 1405, size = 65536} wr = {buf = 0x7ffebd954480 "\026\003\003", pos = 0, used = 0, size = 65536} tls_c = 0x7fa14768f490 enc_rd_buf = <optimized out> n = 0 flush_flags = 0 err_src = <optimized out> ip_buf = '\000' <repeats 63 times> x = <optimized out> tls_dbg = <optimized out> __func__ = "tls_h_read_f" redo_read = <optimized out> continue_ssl_read = <optimized out> __llevel = <optimized out> __kld = <optimized out> __llevel = <optimized out> __kld = <optimized out> #15 0x000055d2519871b4 in tcp_read_headers (c=c@entry=0x7fa1475c0dc0, read_flags=read_flags@entry=0x7ffebd9647d0) at core/tcp_read.c:445 bytes = <optimized out> remaining = <optimized out> p = <optimized out> r = 0x7fa1475c0ee8 mc = <optimized out> body_len = <optimized out> tvnow = {tv_sec = 1742100857, tv_usec = 944923} tvdiff = <optimized out> mfline = <optimized out> mtransid = <optimized out> __func__ = "tcp_read_headers" #16 0x000055d251989cef in tcp_read_req (con=0x7fa1475c0dc0, bytes_read=bytes_read@entry=0x7ffebd9647c8, read_flags=read_flags@entry=0x7ffebd9647d0) at core/tcp_read.c:1509 bytes = <optimized out> total_bytes = 0 resp = 1 size = <optimized out> req = 0x7fa1475c0ee8 dst = {send_sock = 0x55d251a91020, to = {s = {sa_family = 52024, sa_data = "\257Q\322U\000\000\002\000\000\000\376\177\000"}, sin = {sin_family = 52024, sin_port = 20911, sin_addr = {s_addr = 21970}, sin_zero = "\002\000\000\000\376\177\000"}, sin6 = {sin6_family = 52024, sin6_port = 20911, sin6_flowinfo = 21970, sin6_addr = {__in6_u = {__u6_addr8 = "\002\000\000\000\376\177\000\000\030\000\000\000\000\000\000", __u6_addr16 = {2, 0, 32766, 0, 24, 0, 0, 0}, __u6_addr32 = {2, 32766, 24, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 52024, __ss_padding = "\257Q\322U\000\000\002\000\000\000\376\177\000\000\030", '\000' <repeats 23 times>, "\001\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000\262+\226Q\001 \000\000\200,5L\241\177\000\000 \020\251Q\322U\000\000 \020\251Q\322U\000\000\200\303\257Q\322U\000\00009SG\241\177\000\000\330G\226\275\376\177\000\000\b\000\000\000\000\000\000", __ss_align = 20}}, id = 1, send_flags = {f = 1, blst_imask = 0}, proto = 14 '\016', proto_pad0 = 0 '\000', proto_pad1 = 0} c = <optimized out> ret = <optimized out> again = <optimized out> __func__ = "tcp_read_req" #17 0x000055d25198ebec in handle_io (fm=fm@entry=0x7fa14c352c80, events=events@entry=1, idx=idx@entry=-1) at core/tcp_read.c:1906 ret = 8 n = 0 read_flags = RD_CONN_SHORT_READ con = 0x7fa1475c0dc0 s = 14 resp = <optimized out> t = <optimized out> ee = 0x0 __func__ = "handle_io" error = <optimized out> #18 0x000055d25199469b in io_wait_loop_epoll (repeat=repeat@entry=0, t=2, h=0x55d251c49d60 <io_w>) at core/io_wait.h:1075 n = 1 r = 0 fm = 0x7fa14c352c80 revents = 1 __func__ = "io_wait_loop_epoll" #19 0x000055d251994e87 in tcp_receive_loop (unix_sock=<optimized out>) at core/tcp_read.c:2026 __func__ = "tcp_receive_loop" #20 0x000055d25197e72c in tcp_init_children (woneinit=woneinit@entry=0x7ffebd964d1c) at core/tcp_main.c:5391 r = <optimized out> i = <optimized out> reader_fd_1 = 39 pid = <optimized out> si_desc = "tcp receiver (generic)\000\000`M\226\275\376\177\000\000\001\000\000\000\000\000\000\000\210\035\223Q\322U\000\000 \020\251Q\322U\000\000\220\315\257Q\322U\000\000\220\242/L\241\177\000\000\025\000\000\000\000\000\000\000\004\000\000\000\000\000\000\000%\020\251Q\322U\000\000HU\257Q\322U\000\000e\005\000\000\000\000\000\000 \020\251Q\322U\000\000\3750\235Q\322U\000" si = <optimized out> __func__ = "tcp_init_children" error = <optimized out> #21 0x000055d2517c6c7e in main_loop () at main.c:1950 i = <optimized out> pid = <optimized out> si = <optimized out> si_desc = "udp receiver child=7 sock=10.0.90.1:5060\000\000\000\000\322U\000\000?\342\017K\241\177\000\000\022\000\000\000\377\377\377\377\200\006\265L\241\177\000\000\253m\237L\241\177\000\000\361|\003\000\000\000\000\000\t\000\000\000\000\000\000\000Sat Mar 15 05:25:32 2025\n\000\264L\241\177\000" nrprocs = <optimized out> woneinit = 1 __func__ = "main_loop" error = <optimized out> #22 0x000055d2517bc151 in main (argc=<optimized out>, argv=<optimized out>) at main.c:3256 cfg_stream = <optimized out> c = <optimized out> r = <optimized out> tmp = 0x7ffebd966e7f "" tmp_len = 32673 port = 5060 proto = 0 aproto = 0 ahost = 0x0 aport = 0 options = 0x55d251aa74b8 ":f:cm:M:dVIhEeb:B:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 3209119183 rfd = <optimized out> debug_save = <optimized out> debug_flag = <optimized out> dont_fork_cnt = 0 n_lst = <optimized out> p = <optimized out> st = {st_dev = 24, st_ino = 46237, st_nlink = 2, st_mode = 16888, st_uid = 899, st_gid = 899, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1742012732, tv_nsec = 217328715}, st_mtim = {tv_sec = 1742012732, tv_nsec = 217328715}, st_ctim = {tv_sec = 1742012732, tv_nsec = 217328715}, __glibc_reserved = {0, 0, 0}} l1 = <optimized out> tbuf = '\000' <repeats 32 times>, "\030\202\237\275\376\177\000\000@ \002\000\000@\003\000\000\001", '\000' <repeats 23 times>, "@\003\000\000@ \003\000\000@\003\000\000@\003\000\000@\003\000\000@\003\000\000@ \003\000\000@\003\000\000@\003\000\000@\003\000\000@\003\000\000@ \003\000\000@\003\000\000@\003\000\000@\003\000\000@\003\000\000@ \003\000\000@\003\000\000@\003\000\000@\003\000\000@\003\000\000@\003", '\000' <repeats 11 times>, "\001", '\000' <repeats 118 times>... option_index = 12 long_options = {{name = 0x55d251a9230b "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x55d251a937a4 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x55d251a9826a "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x55d251a92310 "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x55d251a92316 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x55d251a9231f "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x55d251a92329 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x55d251a92333 "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x55d251a9233e "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x55d251a92347 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x55d251a938c1 "debug", has_arg = 1, flag = 0x0, val = 1032}, {name = 0x55d251a92352 "cfg-print", has_arg = 0, flag = 0x0, val = 1033}, {name = 0x55d251a9235c "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x55d251a92363 "all-errors", has_arg = 0, flag = 0x0, val = 1035}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main" [Inferior 1 (process 1782236) detached] ---end 1782236 ------------------------------------------------------- ``` Does anybody have an idea what this could be? I never saw that with previous versions, and now we can see it on several instances, but we are yet unable to reproduce. Not sure if this might be related, but we also see many messages like `Mar 17 13:39:03 ttel /usr/sbin/kamailio[2882127]: CRITICAL: <core> [core/tcp_main.c:5558]: tcp_timer_check_connections(): message processing timeout on connection id: 402 (state: 0) - closing` Best regards Mathias

2 6

Kamailio v5.8.6 Released
by Daniel-Constantin Mierla 20 Mar '25

20 Mar '25

Hello, Kamailio SIP Server v5.8.6 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.6. For more details about version 5.8.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/03/kamailio-v5-8-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.8.x is currently the previous stable release series, the latest is 6.0.x and v6.0.1 was released on March 10, 2025. 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 Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Releasing v5.8.6
by Daniel-Constantin Mierla 20 Mar '25

20 Mar '25

Hello, I am considering to release Kamailio v5.8.6 (out of branch 5.8) on Thursday, March 20, 2025. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. The most of the commits are backported, should any be found missing, reply to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 1

Issue with async_cdp_callback() of MAR diameter message
by Minello, Loredana 18 Mar '25

18 Mar '25

Hello, I have an issue during the registration phase with the Diameter MAR procedure. The issue is as follows: 1. I perform a registration. 2. The MAR is correctly sent using the function: ims_www_challenge("REG_MAR_REPLY", "$var(auth_realm)"); 3. Upon receiving the MAA, after executing async_cdp_callback(), the execution does not return to the Kamailio reply route indicated (REG_MAR_REPLY). I tried replacing t_continue with t_continue_skip_timer in the async_cdp_callback() function of the MAR (see below), and everything works correctly. //tmb.t_continue(data->tindex, data->tlabel, data->act); tmb.t_continue_skip_timer(data->tindex, data->tlabel, data->act); free_saved_transaction_data(data); return; Note: When I send the SAR, I do not have this issue, but I noticed that in the corresponding async_cdp_callback(), the t_continue_skip_timer function is used. Do you think the modification I made to async_cdp_callback() for the MAR is correct? Why does t_continue not work? Thanks.

1 0

dispatcher not distributing well?
by David Villasmil 17 Mar '25

17 Mar '25

Hello guys, i have a kamilio with a dispatcher list in db like +------------------------------------------------+----------------------------------------------+----------------------------+ | id | setid | destination | flags | priority | attrs | description | register_datetime | +------+-------+---------------------+-------+----------+------------------------------+---------------+---------------------+ | 3239 | 430 | sip:gateway-08:5080 | 0 | 10 | duid=gateway-08;weight=50 | gateway-08 | 2023-03-21 00:19:44 | | 3246 | 430 | sip:gateway-11:5080 | 0 | 10 | duid=gateway-11;weight=50 | gateway-11 | 2023-05-15 23:00:28 | | 3247 | 430 | sip:gateway-12:5080 | 0 | 10 | duid=gateway-12;weight=50 | gateway-12 | 2023-05-15 23:06:57 | | 3249 | 430 | sip:gateway-14:5080 | 0 | 10 | duid=gateway-14;weight=50 | gateway-14 | 2023-05-15 23:24:44 | | 3276 | 430 | sip:gateway-15:5080 | 0 | 10 | duid=gateway-15;weight=50 | gateway-15 | 2024-05-01 19:43:19 | | 3277 | 430 | sip:gateway-16:5080 | 0 | 10 | duid=gateway-16;weight=50 | gateway-16 | 2024-05-01 19:51:16 | | 3278 | 430 | sip:gateway-1:5080 | 0 | 10 | duid=gateway-17;weight=50 | gateway-17 | 2024-05-01 19:58:13 | | 3279 | 430 | sip:gateway-17:5080 | 0 | 10 | duid=gateway-18;weight=50 | gateway-18 | 2024-05-01 20:06:44 | | 3280 | 430 | sip:gateway-3:5080 | 0 | 10 | duid=gateway-19;weight=50 | gateway-19 | 2024-05-01 20:14:50 | | 3281 | 430 | sip:gateway-2:5080 | 0 | 10 | duid=gateway-20;weight=50 | gateway-20 | 2024-05-01 20:23:25 | +------+-------+---------------------+-------+----------+------------------------------+---------------+---------------------+ and i'm dispatching with if(!ds_select_dst("$var(dispatcher_id)", "10")) { xlog("[DISPATCH]: ds_select_dst FAILED!'\n"); send_reply("404", "No destination"); exit; } still kamailio isn't distributing at all but for some reason is favoring high-description gateways: kamctl rpc dlg.list 5 gateway-20:5080;transport=udp" 2 gateway-19:5080;transport=udp" it's forwarding calls mostly to gateway-20, gateway-19, gateway-18, gateway-17... in descending order... Why is kamailio doing that? i'd expected it to distribute to all based on the call load on each gateway. Thanks all! Regards, David Villasmil email: david.villasmil.work(a)gmail.com

3 4

Recommendations for paid work
by Anthony Ford 14 Mar '25

14 Mar '25

Hi Can anyone recommend someone that can help us upgrade our 2 Kamailio servers please. After several failed attempts to do this ourselves, it is time to call in the professionals. 5.6.0-dev to the latest The current set up is 2 servers working in tandem, redis and mysql plus nebula. We do have a test server also attached, that can be used for testing. Thanks Tony

6 5

add_contact_alias modifies only one Contact address
by James Browne 13 Mar '25

13 Mar '25

- https://kamailio.org/docs/modules/5.8.x/modules/nathelper.html#nathelper.f.… This add_contact_alias() function changes only the first Contact address in a REGISTER request, even though it's normal (RFC3261, 10.2) for a REGISTER to have multiple Contact addresses. This seems to me to be a kamailio bug. For example, a user might remove one binding while adding a second, in a REGISTER request. If kamailio relays this while adding an alias to only the expires=0 binding, then the new binding may be stored incorrectly in the registrar. James

2 4

Server selecting TCP transport
by Jon Bonilla (Manwe) 13 Mar '25

13 Mar '25

Hi all Using 5.8.4 I have a trunk with a permanent registration created by rpc in the location table. It looks like this: kamcmd> ul.lookup location sbc.mydomain.com { AoR: sbc.mydomain.com Contacts: { Contact: { Address: sip:xxx@sbc.mydomain.com Expires: -102739 Q: 1 Call-ID: dfjrewr12386fd6-343(a)kamailio.rpc CSeq: 1 User-Agent: kamailio SIP Router - RPC Server Received: [not set] Path: <sip:192.168.255.202:5060;lr> State: CS_SYNC Flags: 0 CFlags: 0 Socket: [not set] Methods: 0 Ruid: ulcx-67c9e2ff-19f8-1 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 0 Last-Keepalive: 1741899218 KA-Roundtrip: 0 Last-Modified: 0 } } } In this situation the proxy sends the request to 192.168.255.202 which is a kamailio load balancer which is responsible for sending the call to the customer. The problem is that kamailio, when relaying this request, first performs TCP srv lookup and tries to send the request via TCP. There's no transport=tcp in the address. The lb's dns settings are: dns_try_ipv6=no dns_retr_time=1 dns_retr_no=1 dns_use_search_list=no use_dns_cache=on use_dns_failover=on dns_srv_lb=on dns_try_naptr=no dns_sctp_pref=1 dns_tcp_pref=1 dns_tls_pref=1 dns_udp_pref=1 Seems like this kamailio tries TCP SRV first, and then TLS and then UDP if the previous doesn't bring results. But no NAPTR is enabled. I don't know why it will choose TCP over UDP in this case. -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

SDP Truncation When Relaying from WebSocket to UDP – Expected Behavior?
by Pavan Kumar 13 Mar '25

13 Mar '25

Hello Kamailio Community, I recently encountered an interesting behavior when using Kamailio to relay SIP messages from a WebSocket client to FreeSWITCH over UDP. *Observed Behavior:* - The WebSocket client sends an *INVITE* with a large SDP. - Kamailio forwards this request over UDP to FreeSWITCH. - Upon inspection, the *SDP in the relayed message appears truncated*. (Probably MTU limit) - *Surprisingly, the call still establishes successfully*, and there are no noticeable issues with audio or call setup. *My Questions:* 1. *Is this expected behavior?* Should Kamailio automatically truncate SDP when relaying from WebSocket to UDP? 2. *Could this be working accidentally?* For example, is FreeSWITCH handling the partial SDP gracefully by default? 3. *Should I be concerned about potential failures in different scenarios?* (e.g., ICE candidate loss, missing codec negotiation) I would appreciate any insights from the community on whether this is a known or expected behavior in Kamailio, or if it might be a configuration issue. Thanks in advance for your help! Best regards, Pavan Kumar

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Need help using Kamailio IMS and Open5GS for 5G core
by Pete Kay 13 Mar '25

13 Mar '25

Hi I am able to compile Kamailio IMS and Open5GS. The instruction in https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ is for 4G using Diameter. For 5G, we need to use HTTP protocol. Does anyone know if there is any guide on how to configure Kamailio IMS and Open5GS on 5G core using HTTP protocol for PCRF ? Thanks for millions for all your help. Pete

2 1

Re: upgrading process
by Anthony Ford 12 Mar '25

12 Mar '25

Thanks Ben Really appreciate your response Tony From: Ben Kaufman via sr-users <sr-users(a)lists.kamailio.org> Reply to: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Wednesday, 12 March 2025 at 16:20 To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Cc: Ben Kaufman <bkaufman(a)bcmone.com> Subject: [SR-Users] Re: upgrading process Luckily there usually aren't a lot of breaking changes in Kamailio updates, and config files are mostly back compatible. That doesn't mean that there are never changes, but it does mean that the number of issues to resolve is usually small. * wanted to know if I can run make install on every iteration i.e. 5.6.1, the 5.6.2 etc There shouldn't really be any changes needed to config for the patch/fix versions like that (third decimal place) so those should be fine (and is good practice if you're not able to quickly update to a supported version for any reason), but it shouldn't be necessary to touch every incremental version to upgrade Kamailio. Every major release has an upgrade guide from the pervious version, so in your case: * 5.6 to 5.7: https://www.kamailio.org/wikidocs/install/upgrade/5.6.x-to-5.7.0/ * 5.7 to 5.8: https://www.kamailio.org/wikidocs/install/upgrade/5.7.x-to-5.8.0/ * 5.8 to 6.0: https://www.kamailio.org/wikidocs/install/upgrade/5.8.x-to-6.0.0/ It still shouldn't be necessary to touch each individual version while upgrading. I'd recommend reading through these docs in order taking note of the following: * Look for potentially breaking changes in the core. I don't see any in those documents, but a good example would be if upgrading from 4.x to 5.x how fifo was deprecated. * Look for modules changes. Mostly you'll see things like new features which shouldn't be breaking. A few modules were deprecated/archived in 5.8 ( app_sqlang, and auth_identity). Sometimes modules have small breaking changes in them as well, for example in 6.0 the dialog module dropped support for the dlg_flag parameter. Check your kamailio configs to see if any of these listed changes will impact you. * Follow any DB upgrade changes. These are usually simple changes - things like adding an index to a column, or changing a varchar(n) datatype to be varchar(n+y) which won't break your existing config. Another reply correctly points out that you should watch for changes to the version table. This is possibly a situation where a conflict could arise: The version required for one table might be 1 in 5.6 and 2 in 6.0. It might be worth while to copy the existing version table to a new table (like version_60 for example), and use kamailio's version_table parameter to use the new version table on the updated config. (note: The upgrade guides don't show any table version changes, so you probably don't need to do this). Finally, as a recommendation, smoke test your config in docker. You might not have the container setup in a way to actually receive SIP messages, but "does the config file load correctly" is a nice thing to have resolved ahead of time. * Regards, Kaufman ________________________________ From: Anthony Ford via sr-users <sr-users(a)lists.kamailio.org> Sent: Wednesday, March 12, 2025 9:30 AM To: sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org> Cc: Anthony Ford <tony.ford(a)apalo.co.uk> Subject: [SR-Users] upgrading process CAUTION: This email originated from outside the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. Hi I need some help please. I am very inexperienced with Kamailio, and am trying to work out how to upgrade without rewriting any of the current config. I am going from 5.6.0 to 6.0.0 and wanted to know if I can run make install on every iteration i.e. 5.6.1, the 5.6.2 etc. until I need to make a bigger step, and may have to change tables etc. In the document for upgrading 5.6.x to 5.7.x it says make change to Watch table, but I have no watch table. Very confused. Thanks Tony

1 0

CANCEL suspension and forking
by Alex Balashov 12 Mar '25

12 Mar '25

Hi, When a CANCEL is received and matches a known INVITE transaction, it is forked to all existing branches of that INVITE transaction, at least with these settings: modparam("tm", "cancel_b_method", 2) modparam("tm", "failure_reply_mode", 3) I have a mobile endpoint that is slow to be pushed awake and reregister, tsilo-style, and by the time it finally appears on the scene, the underlying INVITE transaction has been terminated. Accordingly, that endpoint receives no CANCEL. You might ask why that's a problem, if it also didn't receive a forked INVITE. The problem is that Apple's CallKit requires interactive presentation of VoIP APNS notifications as though they are "calls", leading to the phenomenon of what might be called "ghost calls" extrinsically to the actual mobile application. Anyway, I am aware of numerous possible solutions to this problem in principle; most involve sending an out-of-band SIP message to signify an alternative cancellation of the call. However, wonder if I am missing some best-practical approach, perhaps through an overlooked TM facility, or some way of keeping the transaction alive longer than the CANCEL from the UAC would ordinarily require? I don't see how that would work, but thought better to ask. Thanks! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

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upgrading process
by Anthony Ford 12 Mar '25

12 Mar '25

Hi I need some help please. I am very inexperienced with Kamailio, and am trying to work out how to upgrade without rewriting any of the current config. I am going from 5.6.0 to 6.0.0 and wanted to know if I can run make install on every iteration i.e. 5.6.1, the 5.6.2 etc. until I need to make a bigger step, and may have to change tables etc. In the document for upgrading 5.6.x to 5.7.x it says make change to Watch table, but I have no watch table. Very confused. Thanks Tony

3 3

Set TTL for skipped NAPTR record due to all IPs are unreachable
by Mathias Schneuwly 11 Mar '25

11 Mar '25

Hi I've a question regarding skipped NAPTR records. In our setup, we are using quite big DNS TTLs (860400). If there is an issue on the target server, the NAPTR record is marked as skipped "Mark to skip _sips._tcp.... NAPTR record due to all IPs are unreachable" until the TTL is expired. Since our TTLs are so big, the destination is no longer reachable during this time, even if the server is reachable again. I've seen that I can set "dns_cache_max_ttl" to reduce the time, but this is of course for all cached DNS records. Is there a way to set how long a NAPTR record is skipped or disable the feature at all? Or maybe skip the NAPTR only for the current dialog or transaction? Thanks for your help Best regards Mathias

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Kamailio v6.0.1 Released
by Daniel-Constantin Mierla 10 Mar '25

10 Mar '25

Hello, Kamailio SIP Server v6.0.1 stable release is out. This is a maintenance release of the latest stable branch, 6.0, that includes fixes since the release of v6.0.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v6.0.x. Deployments running previous v6.0.x versions are strongly recommended to be upgraded to v6.0.1. For more details about version 6.0.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/03/kamailio-v6-0-1-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 Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Releasing v6.0.1
by Daniel-Constantin Mierla 10 Mar '25

10 Mar '25

Hello, I am considering to release Kamailio v6.0.1 (out of branch 60) on Monday, March 10, 2025. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

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Kamailio with Presence Module and Publishing by Asterisk
by Ludovic GRANGE 05 Mar '25

05 Mar '25

Hi, I'm having a problem managing the presence between Kamailio and Asterisk. I have configured Kamailio as a presence server. Asterisk sends presence information via publishing sip request. My softphone which uses Kamailio as a presence server and as a proxy has 2 BLFs configured. On a first call, the presence status is displayed on the 2 keys. But for the 2nd call, only the extension receiving the call has its status changed on the BLF. Via sngrep, the Publishing requests arrive on Kamailio but the DB is not updated because the e-tag does not match. If I wait until the end of the expiry period (60s for the test), the DB is flushed and the 1st call works again. Could you please help me? Thank you Translated with DeepL.com (free version) -- Bien cordialement, Bande verticale Prosoluce Ludovic Grange Administrateur systèmes & réseaux Picto mailludovic.grange(a)prosoluce.fr <mailto:ludovic.grange@prosoluce.fr> Picto tél 05 31 61 62 25 <tel:+33531616225> Picto webwww.prosoluce.fr <https://www.prosoluce.fr> Logo Prosoluce

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Best practices for proxy protocol and TLS offloading
by Alex Balashov 04 Mar '25

04 Mar '25

Hello, I am converting a Kamailio WSS/TLS gateway to pure TCP, with an external HAProxy handling TLS termination and emitting the proxied connection as plain TCP. I am using the PROXYv2 protocol, with `tcp_accept_haproxy=yes`, to convey the upstream network and transport-layer reachability info into Kamailio. I am trying to figure out best practices for mapping the contacts on those connections to the proxied connections themselves. For registrations, tcp_force_alias() (with `tcp_accept_aliases=yes`) works well, and inbound calls go to the right place. However, I'm not quite sure what to do with other flows, such as, for example, in-dialog requests on inbound calls going to the TLS endpoints. There are obviously a lot of possibilities, all or most of which I've tinkered with. These generally involve either {s.replace}-ing `;transport=tls` with `;transport=tcp` in the contacts received from the client, or using the traditional `nathelper` contact alias / RURI alias bag of tricks. However, I don't like the former solution because it leads to a non-compliant R-URI going to the endpoint (it's not the ;transport it sent in its contact), and I don't like the latter because it seems like there are too many moving parts. When tcp_force_alias() works so well for registrations, there must be some small linchpin I'm missing for normal request-reply flows. What is it? Thanks in advance! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

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Topos Module Questions
by David Tek 04 Mar '25

04 Mar '25

Hello, I am trying to configure the Topos module and have some questions. # topos:msg-outgoing vs topos:msg-sending https://www.kamailio.org/docs/modules/5.8.x/modules/topos.html#topos.e.msg_… I am trying to understand the difference between these two event routes(beyond the content of the SIP message ). I notice that both are invoked when OPTIONS are generated by the local Kamailio server but only msg-sending is invoked for messages transiting the Kamailio( ie proxied, not locally generated). Is there some difference in how these event_routers are invoked beyond the content of the SIP message? # tops -> requests/replies When I am in event_route[topos:msg-sending] { and invoke 'drop', does this apply to the current message or does it apply to the transaction? I have a Kamailio node with devices on either side. Let's call side A untrusted and side B trusted. I want TOPOS happening against the untrusted device but not the trusted device. I have found that when I invoke 'drop' on a request coming from A to B, it disables TOPOS not only on that packet, but also the reply comes from B back to A, TOPOS isn't applied when the reply exits Kamailio towards A. So if I call 'drop' on the request coming from A to B, the reply coming from B to A has no TOPOS applied. How do I get TOPOS applied to all traffic facing a particular device while leaving that traffic un-stripped when the same traffic is interacting with other devices that are trusted? Thanks, David

2 2

Torso Sex Toy for Men & Women
by feifeiwangf＠uusextoy.com 04 Mar '25

04 Mar '25

The sex doll torso is a very light mini sex doll. Usually, its weight is 4kg to 20kg. These sex doll torsos are made of TPE or silicone. Of course, they are all very safe materials. https://www.sextorso.com/

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SHA-256 HASH
by Social Boh 04 Mar '25

04 Mar '25

Hello list, is it possible create a user using sha-256 hash? My goal is Asterisk PBX Register ----- sha-256 digest authentication -> Kamailio Regards -- --- I'm SoCIaL, MayBe

2 3

KEMI logging function name output
by Dmitry Sytchev 28 Feb '25

28 Feb '25

Hi all! By default logging from KEMI with KSR.log() is prefixed with something like this: <core> [core/kemi.c:152]: sr_kemi_core_log(): Can we somehow disable this not to print it every time with every message? Thanks! [image: image.png] -- Best regards, Dmitry Sytchev

2 2

SNMP thresholds
by Sergio Charrua 27 Feb '25

27 Feb '25

Hi all! we have set SNMP stats on our production server. The server is only receiving 1% of the calls, less than 1 CAPS, but we are receiving some SNMP alarms that are a bit weird to us. [image: image.png] This is an example. As we were unsure about the thresholds to be set for Msg Queue, we set Minor to 20 and Major to 100, which resulted in some alarms when we exceeded the Major threshold. From our understanding, the Max Major MSG Queue threshold was 2176, which obviously generated an alarm. But we can't understand why when a Msg Queue depth of 1280 generated an Minor Threshold alarm when the threshold is 20!! Why not generate a Major Threshold alarm instead, as it did reach the Major Threshold ? Also, does anyone have a clue on how to estimate Major and Minor thresholds? How does this Msg Queue handler work? Is it Kamailio's level or on OS/Kernel level that it gets the statistics from? Apart from the official Kamailio SNMPStats docs, any doc or article one would share? greatly appreciated. ( I so much hate SNMP!!! :) ) Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

1 0

Tweaking TCP/TLS connection timers
by Ihor Olkhovskyi 27 Feb '25

27 Feb '25

Hello, I'm observing a bit strange behavoir of Kamailio, but maybe I'm wrong in my assumptions. In some cases Kamailio just closes absolutely valid TCP connection, as I see it in Wireshark [image: image.png] At the same time I do see in Kamailio logs a lots of lines with tcp_timer_check_connections(): message processing timeout on connection id: XXXXX (state: 0) - closing I see that many TCP connections are closing (and it's ok as we have mobile devices connected and loosing connections for them is ok) But for me this connection is perfectly ok looking on TCP keepalives, what can cause this behavoir? I have these TCP options 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=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 Kamailio 5.8.3 Any hints are appreciated. -- Best regards, Ihor (Igor)

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Kamailio and TLS-PSK Support
by Юрій Пивоваров 27 Feb '25

27 Feb '25

Dear Kamailio Community, I am currently exploring the possibility of configuring Kamailio to use TLS-PSK (Pre-Shared Key) for secure SIP signaling. Has anyone successfully configured Kamailio to work with TLS-PSK? If so, could you share any guidance, configuration examples, or relevant modules that need to be adjusted? In this setup, the SIP client will be a device with very limited resources, making it impossible to use certificates. This is why I am specifically looking into a TLS-PSK solution. I have also tried configuring stunnel as a proxy and set listen = tcp:127.0.0.1:5061 in kamailio config, but I haven't been able to get it working. Any suggestions on what might be missing or misconfigured? Thank you in advance for your help!

1 0

Drop RTP Flow and move to another point
by Social Boh 26 Feb '25

26 Feb '25

Hello List, before spending much time on solving this scenario, I'd like to know if this kind of operation can be done using some Kamailio function (maybe dlg_req_within?) or other method. The scenario is: a B2B-1 server call two different destinations, one is a SIP trunk and another is another B2B-2. Both calls go through Kamailio which uses RTPEngine for the media flow. The idea is to drop the first B2B from RTP Flow (leave for signaling) when both calls are answered and leave everything to Kamailio-RTPEngine, like: LegA: Kamailio-RTPEngine -> B2B-2 LegB: Kamailio-RTPEngine -> SIP trunk Thank you in Advance Regards --- I'm SoCIaL, MayBe

1 1

PRACK Retransmission and 408 Timeout Issue in Kamailio
by 高秉寬 26 Feb '25

26 Feb '25

Dear Kamailio Developers, I am using Kamailio as a SIP server to facilitate a phone call, but I am encountering an issue where the call cannot be established due to Kamailio sending a 408 Request Timeout to the caller. Issue Details: - A (Caller) sends an initial INVITE to Kamailio, which then forwards it to B (Callee). The INVITE contains the following Supported header: Supported: timer,100rel,replaces,precondition,from-change,histinfo,tdialog I suspect the *precondition* option may be relevant to the issue. - *B* responds with a *183 Session Progress* message, which includes: Require: 100rel,precondition Kamailio correctly forwards this response to *A.* - *A* then sends a *PRACK* request to Kamailio. However, this request contains: Require: sec-agree Supported: timer Notably, *precondition* is absent from both the *Require* and *Supported* headers. - Kamailio forwards this *PRACK* to *B*, and *B* responds with *200 OK*, including: Supported: 100rel,from-change,tdialog - However, Kamailio does not seem to acknowledge *B's 200 OK* and instead *keeps retransmitting the PRACK* to *B*. B responds with *200 OK* again, but the cycle repeats. - After three retransmissions of *PRACK*, Kamailio eventually sends *408 Request Timeout* to *A*, leading *A* to send a *CANCEL* request. Device Used: - *Caller (A):* Asus ROG Phone 5 - *Callee (B):* Asus ROG Phone 6 Both are commercially available mobile phones. Possible Cause: According to *3GPP TS 24.229, Subclause 5.1.3.1*, If the received response with an SDP body includes a Require header with > the "precondition" option-tag, the originating UE shall include a Require > header with the "precondition" option-tag in responses with an SDP body to > subsequent requests that include an SDP body and include the "precondition" > option-tag in either the Supported or Require header field received > in-dialog. In this case: - The *183 Session Progress* from *B* contains Require: *precondition*. - The subsequent *PRACK* from *A* does not include *precondition* in either *Require* or *Supported*. Could this mismatch be the reason Kamailio keeps retransmitting *PRACK* and ignoring *B's 200 OK*? Or could there be another factor causing Kamailio to behave this way? If additional details are needed, please let me know. Best regards, Ping Kuan Kao

3 4

Issue with registrations
by Anthony Ford 26 Feb '25

26 Feb '25

Hi Version = 5.6.0-dev0 We are seeking help to resolve an issue with registering devices to 3CX (All 3CX servers, have tried 8 different ones) . all the auth info including passwords etc are identical on the 3cx servers. We use Kamailio to register mobile sim cards to 3rd party sip servers i.e. 3CX. We send the normal Username, server addresses and auth details but are experiencing failures with 3CX rejecting the registration and return Proxy Auth failure 407. We have noticed in the pcaps from both sides that the expire value is changing from 180 as an example to 0. As an example In the uac reg_dump the values are : l_uuid: +447529135850<tel:+447529135850> l_username: +447529135850<tel:+447529135850> l_domain: sbc-mt.asim.apalo.io<http://sbc-mt.asim.apalo.io/>;transport=udp r_username: 1001 r_domain: egte.3cx.uk<http://egte.3cx.uk/> realm: <null string> auth_username: FdlHDC1IvA auth_password: extracted auth_ha1: none auth_proxy: sip:egte.3cx.uk:5060 expires: 60 flags: 0 diff_expires: -1738853736 timer_expires: 0 reg_init: 1738850836 reg_delay: 0 contact_addr: k01.asim.apalo.io<http://k01.asim.apalo.io/>;transport=tcp socket: <null string> [cid:image001.png@01DB86AB.2701F190] [cid:image002.png@01DB86AB.2701F190] [cid:image003.png@01DB86AB.2701F190] Looking forward to hearing from you. Many thanks Tony Ford

3 3

REG_MAR_REPLY never get executed
by stinktohighheaven＠gmail.com 26 Feb '25

26 Feb '25

Using Kamailio v5.3, based on the example for scscf found here: https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/scscf/ka… after calling ims_www_challenge("REG_MAR_REPLY", "$var(realm)", "$var(alg)"); the control never goes to REG_MAR_REPLY route block, but it goes to the start of the requet_route block. here is the kamailio.cfg script: request_route { ....... /* Handle Registrations */ if (is_method("REGISTER")) { route(REGISTER); exit; } ...... } /** * Handling of REGISTER requests */ route[REGISTER] { ...... #!ifdef WITH_AUTH if (!ims_www_authenticate("$var(realm)")) { #!else if ($var(alg) == "3GPP-Digest" && !ims_www_authenticate("$var(realm)")) { #!endif if ($? == -2) { send_reply("403", "Authentication Failed"); exit; } else if ($? == -3) { send_reply("400", "Bad Request"); exit; } else if ($? == -9) { xdbg("Authentication re-sync requested\n"); ims_www_resync_auth("REGISTER_RESYNC_REPLY", "$var(realm)"); exit; } else { /* user has not been authenticated. Lets send a challenge via 401 Unauthorized */ xdbg("About to challenge! auth_ims\n"); ims_www_challenge("REG_MAR_REPLY", "$var(realm)", "$var(alg)"); exit; } } else { xdbg("Auth succeeded\n"); /* We need to check if this user is registered or not */ if (!impu_registered("location")) { save("PRE_REGISTER_SAR_REPLY", "location"); exit; } else { isc_match_filter_reg("1", "location"); save("REGISTER_SAR_REPLY", "location"); exit; } } } } /** * Handling of REGISTER MAR replies */ route[REG_MAR_REPLY] { xlog("L_DBG","maa_return code is $avp(s:maa_return_code)\n"); /* this is async so to know status we have to check the reply avp */ switch ($avp(s:maa_return_code)) { case 1: /* success */ xdbg("MAR success - 401/407 response sent from module\n"); break; case -1: /* failure */ xerr("MAR failure - error response sent from module\n"); break; case -2: /* error */ xerr("MAR error - sending error response now\n"); send_reply("503", "MAR failed"); break; default: xerr("Unknown return code from MAR, value is [$avp(s:maa_return_code)]\n"); send_reply("503", "Unknown response code from MAR"); break; } exit; } and here's the log: 12(18) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/usr/local/etc/kamailio/kamailio.cfg] l=757 a=27 n=ims_www_challenge 12(18) DEBUG: ims_auth [authorize.c:293]: challenge(): Looking for route block [REG_MAR_REPLY] 12(18) INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [-2] 12(18) DEBUG: ims_auth [authorize.c:317]: challenge(): Need to challenge for realm [ims.mnc071.mcc432.3gppnetwork.org] 12(18) DEBUG: ims_auth [authorize.c:324]: challenge(): Checking if REGISTER is authorized for realm [ims.mnc071.mcc432.3gppnetwork.org]... 12(18) DEBUG: ims_auth [authorize.c:1449]: get_auth_userdata(): Searching auth_userdata for IMPU sip:432710000010044@ims.mnc071.mcc432.3gppnetwork.org (Hash 425) 12(18) DEBUG: ims_auth [authorize.c:460]: challenge(): Suspending SIP TM transaction 12(18) DEBUG: ims_auth [authorize.c:1529]: multimedia_auth_request(): Sending MAR 12(18) DEBUG: ims_auth [cxdx_mar.c:574]: cxdx_send_mar(): Successfully sent async diameter 70(76) INFO: ims_auth [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp (avp_code = 297, vendor_id = 0) 70(76) DEBUG: ims_auth [authorize.c:1303]: new_auth_vector(): new auth-vector with ck [d2deb48b5b3b614ba9112396e3c036fe] with status 0 70(76) DEBUG: ims_auth [authorize.c:1560]: pack_challenge(): setting QOP str used is [, qop="auth"] 70(76) DEBUG: ims_auth [authorize.c:1562]: pack_challenge(): QOP str used is [, qop="auth"] 70(76) DEBUG: ims_auth [authorize.c:1449]: get_auth_userdata(): Searching auth_userdata for IMPU sip:432710000010044@ims.mnc071.mcc432.3gppnetwork.org (Hash 425) 70(76) DEBUG: ims_auth [authorize.c:1458]: get_auth_userdata(): Found auth_userdata 70(76) DEBUG: ims_auth [authorize.c:1682]: add_auth_vector(): Adding auth_vector (status 1) for IMPU sip:432710000010044@ims.mnc071.mcc432.3gppnetwork.org / IMPI 432710000010044(a)ims.mnc071.mcc432.3gppnetwork.org (Hash 425) 70(76) DEBUG: ims_auth [cxdx_mar.c:486]: async_cdp_callback(): DBG:UAR Async CDP callback: ... Done resuming transaction 70(76) INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [1] 70(76) ERROR: tm [t_suspend.c:197]: t_continue_helper(): active transaction not found 70(76) DEBUG: ims_auth [cxdx_mar.c:87]: free_saved_transaction_data(): Freeing saved transaction data: async 11(17) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=380 a=5 n=route

4 8

tm stats on prometheus
by Alexis Fidalgo 25 Feb '25

25 Feb '25

Hello all, i found that tm stats (kamcmd tm.stats) are not present in the response of xhttp_prom (prometheus), so in new configurations that we applied, we have no information/stats on reply codes. Is there a way to add this information to the xhttp_prom response? thanks in advance. regards

2 4

debug memory usage by AVP
by Sergey Safarov 25 Feb '25

25 Feb '25

In the "kamcmd mod.stats all shm" output i see to big value for "create_avp" metric Module: core { tcpconn_new(1205): 35104 counters_prefork_init(211): 61952 create_avp(178): 20737248 cfg_clone_str(132): 304 cfg_shmize(221): 1248 main_loop(1381): 16 Is possible to print AVP list and APV content without kamailio config change and restart?

1 0

Kamailio uses an ephemeral IP port instead of the port defined in $fs
by Leonid Fainshtein 24 Feb '25

24 Feb '25

Hi, I am using Kamailio v.5.8.4 on Ubuntu 24.04, kernel 6.8.0-51-generic In the configuration script I explicitly set $fs="udp:adrr:port". In normal situations, it works properly and I see that the UDP SIP requests are sent from the requested IP and port. Due to some reasons, free space on the server root filesystem was exhausted. I cleaned up the file system and everything looked good except the fact that Kamailio started to send messages from ephemeral IP ports instead of the port defined in $fs. Restart of Kamailio didn't solve the problem. But the server reboot fixed the issue. I don't understand how the lack of free disk space can cause such a problem and I'm afraid that there is another reason that I am missing. Has anybody seen such a problem? Best regards, Leonid

3 3

Outdated location information on deleted registrations
by Mathias Schneuwly 19 Feb '25

19 Feb '25

Hi I'm using Kamailio 5.8.4 and the python KEMI module. When I move the registration from one device to another, I can see for a while two entries with 'kamctl ul show' where the Expires parameter of the "old" registration is set to "deleted" and the "new" registration shows the seconds until the registration expires. "Address": "sips:200101@10.0.1.130:42323 ;transport=tls", "Expires": "deleted", -- "Address": "sips:200101@10.0.1.129:56374 ;transport=tls", "Expires": 3600, So far so good. When I try to fetch the actual registration address using "registrar.reg_fetch_contacts" or "registrar.lookup_xavp", I'm always getting the "old" deleted registration address instead of the new address as long as the deleted record is not removed. Feb 18 10:05:51 ttel /usr/sbin/kamailio[2356327]: WARNING: {1 511 INVITE 952600514-5060-52(a)BA.A.C.BCI} <core> [core/kemi.c:157]: sr_kemi_core_log(): reg_fetch_contacts: sips:200101@10.0.1.130:42323;transport=tls Feb 18 10:05:51 ttel /usr/sbin/kamailio[2356327]: WARNING: {1 511 INVITE 952600514-5060-52(a)BA.A.C.BCI} <core> [core/kemi.c:157]: sr_kemi_core_log(): lookup_xavp: aor=200101, uri=sips:200101@10.0.1.130:42323;transport=tls, socket=tls:10.0.1.1:5061, dsturi=None I also see that the database is only updated after the deleted entry is removed. Until then, the database still contains the old address. Btw, I'm using "modparam("usrloc", "db_mode", 2)", which would explain the delayed persisting into the database. 62|uloc-67b44aae-23f48e-2|200101||sips:200101@10.0.1.130:42323 ;transport=tls|||2460724.91858218|-1.0|482845299-5060-1(a)BA.A.B.BDA|2002|2460724.87691551|0|0|Grandstream GXV3350 1.0.3.52|tls:10.0.1.1:5061 |7135|<urn:uuid:00000000-0000-1000-8000-C074AD188038>|1|0|52|0|16 I'm asking myself whether the behavior is correct that "registrar.reg_fetch_contacts" and "registrar.lookup_xavp" are returning the deleted information instead of the "new" registration information? If yes, is there another way how I could get the "new" registration information? I'm not really interested in the old outdated information... Best regards Mathias

2 2

Info: maintenance work for kamailio.org server on Tuesday, Feb 18, 2025
by Daniel-Constantin Mierla 18 Feb '25

18 Feb '25

Hello, there is a planned mainenance work on the infrastructure for the server hosting kamailio.org services (among them: the main website, wiki, mailing lists) on next week, Tuesday, February 18, 2025, to start about 10:00am Berlin time (CET). Expect some intervals of unavailability, we hope they will be short and most of them not noticeable for the majority accessing these services during the day. Once the work is finished, an announcement will be sent out. The apt repository, github project and the matrix chat channels should not be affected, they can be accessed as usual and can be used a backup communication channels if anyone observes something that has to be addressed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 2

Asterisk ConfBridge behind Kamailio+Rtpengine
by Rahul mm 18 Feb '25

18 Feb '25

Dear all, Thanks for the amazing product ! I’m trying my hands on a scenario involving Asterisk for video_sfu and a bunch of webrtc and sip clients. The condition is that every call must essentially be a conference call for the participants to add more dynamically. I front ended asterisk with Kamailio and RTPEngine for ssl offloading, load balancing, and registrations. So when one user is present in the conference (his invite SDP containing one audio and one video line) i get local video and MoH back, but when another user joins in and a re-invite is send from asterisk containing two video lines - one for the sendrecv and other for send only, rtpengine doesn’t process the sendonly video line - keeps it as is RTP/AVP instead of making it *UDP/TLS/RTP/SAVPF.* *Is there a way to achieve that? or I’m on the wrong path ?* BR, RM

1 0

Kamailio - asterisk registration issue & transparent proxy
by Enzo Damato 17 Feb '25

17 Feb '25

Hi. I'm working on setting up kamailio as a load balancer for some asterisk instances. I've build a configuration that is working for the most part, but I am encountering issues with calls from the asterisk instances to users who's registrations were processed through the kamailio instance. When these users register, asterisk saves thier AOR as the address of the kamailio instance, and then subsequently tires to send invites to kamailio, instead of to the UAC. Since kamailio just proxied the registration, without actually registering the user, it doesn't know what to do with the invite, and the call fails. I would like to configure kamailio to rewrite the sip requests so that all traffic appears to be coming from the end user, rather than from the kamailio instance, essentially making the proxy transparent. Is this possible? Thank you, Enzo Damato

4 6

Prioritize specific codecs
by Temilolu Fayomi 17 Feb '25

17 Feb '25

Hello, I am working with inbound calls on kamailio (specifically fax), and was wondering if there was a way to re-prioritize the codecs found in the sdp of an invite. I've looked at using the sdpops module, but haven't been able to re-arrange the priority of the codecs listed. I've tried removing codecs to help incoming faxes switch over to t38 per my upstream providers instructions (calls will do a t38 re-invite if the g729a is the highest priority). I am just looking for which direction to go from here or if I misunderstand the nature of faxing and kamailio. Thank you in advance, Temilolu Fayomi

3 3

Constant crashes across versions
by Jon Bonilla (Manwe) 17 Feb '25

17 Feb '25

Hi all I have a setup of several proxies behind a load balancer and all of them have several coredumps every day. I've tried versions 5.3, 5.5 and 5.8 and all of them crash I didn't have dbgsym packages installed but now I've built a 5.8.4 version in one of them and can gdb the coredumps. Can't see anything particular and I'm wondering if I'm doing anything wrong. I'm descompressing the coredump with lz4cat. Sometimes I see that the system generates 3 coredumps at the same time. I've tried running gdb in one of them and exec "bt full" I usually see "can't access memory" in some of them but inb others I don't see anything relevant. I'm attaching one coredump. I don't even know if I'm doing it properly. Could you please guide me to how to debug what's going on? thanks

2 4

Info: sipexer v1.2.0 released
by Daniel-Constantin Mierla 11 Feb '25

11 Feb '25

Hello, a short note to announce that that sipexer v1.2.0 was released: - https://github.com/miconda/sipexer/releases/tag/v1.2.0 It is a modern and flexible SIP testing cli tool that can be used to send SIP signalling messages over UDP/TCP/TLS/WebSocket. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

DNS Unlinked Warnings
by John Larsen 10 Feb '25

10 Feb '25

Hi Everyone, We recently attempted an upgrade from kamailio 5.5.7 to 5.7.6. We immediately started to receive a log of warning logs similar to `dns_hash_put_shm_unsafe(): unlinked item 0x7f08799dcd90` The code shows that this warning log was added to the DNS cache in september 2023 - but its unclear what it means to have an unlinked item exactly? Is anyone able to shed any light on how DNS entries might become unlinked, and what the implications of it are? thanks John Larsen John Larsen Tech Lead EJohn.Larsen(a)symbio.global W symbio.global This communication is intended only for the person to whom it is addressed and may contain confidential material. If you received this communication in error, please inform the sender immediately and delete all copies. Please think of the environment before printing this email.

2 1

switch between "sips:" and "sip:" schema
by James Lipski 10 Feb '25

10 Feb '25

Hello, I have a kamailio installation where it's basically doing transport protocol translation. To go into more detail, I have a UA communicating over TLS towards kamailio/ and kamailio forwards the request to a freeswitch installation via UDP, so call-flow is as follows: UA <-- (TLS) --> Kamailio <-- (UDP) --> Freeswitch For the most part, this does work/ it's only when I use a UA that uses "sips:" as the schema instead of "sip:" where issues occur -- in my current topology, freeswitch has some weird behavior where it doesn't disconnect the call properly. At least based on my testing, I believe the issue is relating to the schema -- as the UA in question does have the ability to switch between schemas/ when using "sip:" instead of "sips:" over TLS, no issues with the same call-flow. In the event that I do come across a UA that doesn't have that ability, I was wondering if kamailio has the ability to switch schemas when it sends traffic over the UDP connection. I'm able to replace most aspects of the packet with "sip:" however the only header which contains the "sips:" schema is the contact header. With this alone, it propagates to the B-leg side (UDP side) of the call; and will cause freeswitch to not disconnect calls properly. I'm using the topos module to obfuscate my topology, not sure if that matters. -------- For example, on the A-leg side of the call, kamailio receives this: INVITE sips:12125551000@dev-sip.server.com:5061 SIP/2.0 Via: SIP/2.0/TLS 172.16.1.194:5061;branch=z9hG4bK236557042;rport;alias From: <sips:100000000@dev-sip.server.com>;tag=444048970 To: <sips:12125551000@dev-sip.server.com> Call-ID: 1326419916-5061-4(a)BHC.BG.B.BJE CSeq: 21 INVITE Contact: <sips:100000000@172.16.1.194:5061;transport=tls> Proxy-Authorization: Digest username="100000000", realm="dev-sip.server.com", nonce="Z6Ig0WeiH6Vdxtvej0I3XGCTUAEfh+0/", [uri="sips:12125551000@dev-sip.server.com](mailto:uri=)", response="43c1487d077db36a70fffdd1e25b3b52", algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream HT812 1.0.57.1 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 1034 v=0 o=100000000 8000 8000 IN IP4 172.16.1.194 s=SIP Call c=IN IP4 172.16.1.194 t=0 0 m=audio 5004 RTP/SAVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=crypto:1 AEAD_AES_256_GCM inline:UlX3PAjNBP/aQwtwyf5ymsfJciB+VdFxMElO5zLVooUqmcEzZ8YyQQk9stI=|2^32 ------------------------------------------------------ On the B-leg side of the call, kamailio forwards the following to freeswitch: INVITE sip:12125551000@fs.server.com SIP/2.0 Via: SIP/2.0/UDP 10.64.52.113:5080;branch=z9hG4bK1552.89ec9666294034a5611f908f2a6a427e.0;i=f From: "Test" <sip:16461234567@dev-sip.server.com>;tag=444048970 To: <sip:12125551000@fs.server.com> Call-ID: 1326419916-5061-4(a)BHC.BG.B.BJE CSeq: 21 INVITE Max-Forwards: 15 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 316 Contact: <sips:btpsh-67a21ca2-2dbe86-1@10.64.52.113:5080> v=0 o=100000000 8000 8000 IN IP4 10.64.52.113 s=SIP Call c=IN IP4 10.64.52.113 t=0 0 m=audio 13060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=ptime:20 -------------------------------------------------- As with the above, I was wondering if there is a way within kamailio to switch schemas on the b-leg side of this call example; specifically is there a way to update the contact header so it doesn't use the "sips:" schema when it traverses the UDP connection. Thank you for you time.

4 5

Auth_ephemeral module setup
by Anoop L 04 Feb '25

04 Feb '25

Hi, I'm trying to understand and setup Auth_ephemeral for authentication. My goal is to authenticate users dynamically using an web service for that I bumped into auth_ephemeral module. In the module how is a request made to the web service? Are username and password fetched from Pseudo-Variables variable? Is this module to be worked with xhttp module for http requests? Any help is much appreciated.

1 0

Auth_ephemeral Module Setup
by anoop96115＠outlook.com 04 Feb '25

04 Feb '25

Hi, I'm trying to understand and setup Auth_ephemeral for authentication. My goal is to authenticate users dynamically using an web service for that I bumped into auth_ephemeral module. In the module how is a request made to the web service? Are username and password fetched from Pseudo-Variables variable? Is this module to be worked with xhttp module for http requests? Any help is much appreciated.

1 0

Testing Kamailio IMS Modules with IPSec as P-CSCF
by Saeed Mogharian 03 Feb '25

03 Feb '25

Hi, I’m looking for suggestions on how to test Kamailio IMS modules, specifically for the IMS registration process over IPSec when Kamailio is configured as the P-CSCF node. Are there any scripts or IMS clients available that can help with this? Thanks in advance!

2 1

Kamailio 6.0 presentation at FOSDEM 2025
by Henning Westerholt 31 Jan '25

31 Jan '25

Hello, just a quick announcement, that tomorrow afternoon will be a presentation about Kamailio 6.0 at the FOSDEM conference in Brussels, Belgium [1]. Usually all the talks in this room are streamed online as well. I am happy also to meet during the conference to chat about Kamailio or other topics, just send me a (direct) message. Best regards, Henning [1] https://fosdem.org/2025/schedule/event/fosdem-2025-5978-kamailio-6-0-develo… -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Strange memory stats
by Barry Flanagan 30 Jan '25

30 Jan '25

Hi, I have a 5.8.4 runing on Bookworm that after a while reports very strange memory stats like this: shmem.fragments: '729' shmem.free_size: '18446744009887078560' shmem.max_used_size: '67043934736' shmem.real_used_size: '67043698528' shmem.total_size: '3221225472' shmem.used_size: '67034971568' The settings are: SHM_MEMORY=3072 PKG_MEMORY=256 The shmem.free_size and max_used/max_size seem quite wrong. The output of "mod.stats all shm" shows the below: - 'Module: jsonrpcs' - Total: 0 - 'Module: core' - Total: 7891296 - 'Module: rr' - Total: 0 - 'Module: dialog' - Total: 146144 - 'Module: permissions' - Total: 130672 - 'Module: auth' - Total: 0 - 'Module: topoh' - Total: 0 - 'Module: sl' - Total: 14048 - 'Module: siptrace' - Total: 79184 - 'Module: sqlops' - Total: 0 - 'Module: acc' - Total: 0 - 'Module: xlog' - Total: 0 - 'Module: dmq' - Total: 1520 - 'Module: avpops' - Total: 0 - 'Module: rtpengine' - Total: 69328 - 'Module: db_mysql' - Total: 0 - 'Module: geoip2' - Total: 32 - 'Module: pv' - Total: 0 - 'Module: registrar' - Total: 0 - 'Module: textops' - Total: 0 - 'Module: htable' - Total: 6375584 - 'Module: tsilo' - Total: 74048 - 'Module: usrloc' - Total: 554240 - 'Module: tm' - Total: 6812224 - 'Module: websocket' - Total: 2551536 - 'Module: pike' - Total: 6880 - 'Module: tls' - Total: 5462960 - 'Module: tmx' - Total: 21968 - 'Module: kex' - Total: 3536 - 'Module: cfgutils' - Total: 128 - 'Module: sipdump' - Total: 32 - 'Module: domain' - Total: 5920 - 'Module: dispatcher' - Total: 20144 Thanks! -Barry

2 8

Kamailio v6.0.0 Released - new major version is out
by Daniel-Constantin Mierla 29 Jan '25

29 Jan '25

Kamailio v6.0.0 is out – it comes with 5 new modules and important updates to the core components and existing modules. The increment of the first version number reflects several important updates, among them: multi-threaded option for UDP receiving, option for specifying a range of ports to listen to, support for expressions with variables for most of the module functions parameters, internal libraries are directly inked to the core, support to use cmake to compile and install Kamailio. You can read a bit more detailed release notes at: * https://www.kamailio.org/w/kamailio-v6.0-0-release-notes/ Many thanks to all developers and community members that made possible this release. v6.0.0 brings more flexibility and optimizations across many existing components and modules, new parameters, functions, variables and transformations. Enjoy Kamailio v6.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

2 2

Releasing v5.8.5
by Daniel-Constantin Mierla 29 Jan '25

29 Jan '25

Hello, I am considering to release Kamailio v5.8.5 (out of branch 5.8) on Thursday, Jan 23, 2025. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 2

MS Teams BYE routing
by Barry Flanagan 28 Jan '25

28 Jan '25

Hi all, I have a Teams gw working apart from one issue., In a call originated in Teams,. if the called party hangs up, the BYE is sent to api-du-b-euwe.pstnhub.microsoft.com:443 which does not work. This is taken from the Contact header of the ACK I presume. Here is the ACK to the 200 OK: 28/01/2025 17:08:26.560 +00:00: 52.114.75.24:53728 -> 100.100.198.241:5061 ACK sip:9000@teams.domain.com:5061;transport=tls;tps=atpsh-67990f01-8ed8-1 SIP/2.0 FROM: "Teams User" <sip:+35314375362@sip.pstnhub.microsoft.com:5061;user=phone>;tag=c29cb7a79ddf40f9ba909f3f1034a770 TO: <sip:9000@teams.domain.com:5061>;user=phone;tag=as6e287620 CSEQ: 1 ACK CALL-ID: e49f7df631ee576286ecf159e81952fe MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK46d28f33 CONTACT: <sip:api-du-b-euwe.pstnhub.microsoft.com:443;x-i=0781e8a1-91e6-4f6a-bc4a-73d6890827d4;x-c=e49f7df631ee576286ecf159e81952fe/d/8/703117d8337e4975a8216a54f59035c2> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2025.1.20.11 i.EUWE.7 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY ..and here is the BYE. 28/01/2025 17:08:36.119 +00:00: 100.100.198.241:5060 -> 52.123.134.13:443 BYE sip:api-du-b-euwe.pstnhub.microsoft.com:443;x-i=0781e8a1-91e6-4f6a-bc4a-73d6890827d4;x-c=e49f7df631ee576286ecf159e81952fe/d/8/703117d8337e4975a8216a54f59035c2 SIP/2.0 Record-Route: <sip:teams.domain.com:5061;transport=tls;ftag=as6e287620;lr;r2=on> Record-Route: <sip:100.100.198.241:5060;ftag=as6e287620;lr;r2=on> Via: SIP/2.0/UDP 100.100.198.241;branch=z9hG4bKc89c.0c1cf04a771c30293432e5f1f59331c8.0 Via: SIP/2.0/UDP 102.102.139.182:5060;rport=5060;branch=z9hG4bK3c42c163 Max-Forwards: 69 From: <sip:9000@teams.domain.com:5061>;user=phone;tag=as6e287620 To: "Teams User"<sip:+35314375362@sip.pstnhub.microsoft.com:5061;user=phone>;tag=c29cb7a79ddf40f9ba909f3f1034a770 Call-ID: e49f7df631ee576286ecf159e81952fe CSeq: 102 BYE Content-Length: 0 How can I get around this? Using a pretty standard WITHINDLG route. I am using topos but not manipulating the Contact header. Thanks! -Barry

4 8

RTPengine module - pause/resume recording
by Who AmI 28 Jan '25

28 Jan '25

Hi list, I have been looking at the recording daemon in RTPengine with Kamailio controlling this via rtpengine_manage() flags: "record-call=on" in the initial Invite, or 200OK based on direction. I tested and this works brilliantly for the full calls, but when looking at pause/resume the module documentation is bare and I am a bit confused. It looks like the ngcp docs and Opensips module docs (for RTPengine) describe the pause/resume as just another start/stop flag. Use case is simple, all calls are recorded, but we want the option to pause, then resume once any private or personal information exchange has completed. My question is also simple - Is this the recommended way to pause and then resume recordings or have I missed something? The Kamailio documentation doesn't explicitly say this ( https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html#rtpengi…) so just looking for advice if this is the best route to go. Thanks, John.

2 2

Re: [sr-dev] Re: Branching 6.0 series
by Daniel-Constantin Mierla 28 Jan '25

28 Jan '25

Hello, On 20.01.25 08:25, Daniel-Constantin Mierla via sr-dev wrote: > Hello, > > I think we should branch it on Wednesday, Jan 22, 2025, no matter on > what stage we are with cmake support. The old-Makefiles should be kept > anyhow (as they are or in a special folder with an easy way to recover > them), because there are many bits and pieces that can be discovered > later when more users. > > Because there were no many bugs reported to the C code specific to 6.0, > we can aim releasing on January 29, 2025, with old-Makefile still to be > used for tasks that are not yet covered by cmake. shall we still aim for releasing v6.0.0 on this upcoming Wednesday, January 29, 2025? We the old-Makefiles kept, it is doable, the wiki pages needs to be created for this release series, which I can do tomorrow, as I waited to see if how it should be the recommended method for compilation/installation. If the feeling is that we should wait, then probably the target date has to be shifted somewhere in the week starting on February 10, 2025, because Fosdem follows this weekend and then some countries have a week holiday break. If no other opinions and change of decision, the release will be done on Wednesday, with content using the old-Makefiles (to reuse existing tutorials) and reference to cmake variant when it is the case. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

2 1

defining configuration variables
by Sergio Charrua 27 Jan '25

27 Jan '25

Hello, just wondering if there is any way to define 2 constants as : # sets quarantine period the URL is set, in secs, after number of HTTP_RETRIES retries #!define SOME_VAR_1 60 # sets the configuration values for HTable module parameter #!define SOME_VAR_2 "some_configration_value" + SOME_VAR_1 I couldn't find anything in docs, and made several different ways without success. Any suggestions? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

2 2

UDP worker lock investigation
by Володимир Іванець 26 Jan '25

26 Jan '25

Hello everyone! I was wondering if it is possible to get information about a packet that is being possessed by the worker without increasing log level? I have an installation with Kamailio that is configured with two interfaces. Once in a while Kamailio stops accepting requests on one of the interfaces. It works fine for requests from other one. I think that worker processes might be locked while processing some requests. Enabling debug log is not an option because the traffic is quite high and the log file will not just be enormous, some lines will be missing. It looks like syslog will skip some of the log entries. So is it possible to check what packet worker process is currently processing? Or do I need to add some log entries with lower log level to trace requests manually? Thank you!

3 4

How to access registration contact user agent string in branch route after location lookup?
by Benoit Panizzon 24 Jan '25

24 Jan '25

Hi List I have some CPE which require special handling like removing 100rel and update from the supported methods because they misbehave. Previously, I was using reg_fetch_contacts and then appending the branches manually and stacking the UA in an AVP. With the branch index this allowed me to get the AVP containing the UA to which the call was branched in the branch route and perform filtering of methods based on UA, but it was complicated. Now I use the intended function lookup_to_dset() but now I miss how to access the register contact UA from within a branch route. Is there an intended way to access that information? 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 ______________________________________________________

3 7

Use local_identity or custom data to set callerID
by matu lopi 23 Jan '25

23 Jan '25

Hi team, Is it possible to use either local_identity or custom data properties to define the CallerID during the RTCSession lifecycle? Thank you.

2 1

Issue with ACK on INVITE
by Barry Flanagan 23 Jan '25

23 Jan '25

Hi folks, Trying to hook up a new endpoint but am having an issue. Kamailio is in front of an Asterisk box. They send an INVITE, we send 100, then 200 OK. However, when they send their ACK, the RURI is not set to the Contact of the 200, instead it is<number>@<proxy ip>. This causes the ACK to get routed to the proxy itself and the call fails. 82.160.190.100:5060 = Kamailio IP Port 198.201.240.242:5080 = Asterisk IP/Port 182.33.174.10:5060 = Provider endpoint The far endpoint say they cannot fix the RURI - should I be able to handle this ACK below? My understanding is the ACK's RURI should be the Contact of the 200 OK. 200 OK sent from us to the Provider (Contact shows correct URI) ========================================================= 2025/01/15 17:14:49.470634 82.160.190.100:5060 -> 182.33.174.10:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.0 Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK5bbf9591;rport=5060 Record-Route: <sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Record-Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Record-Route: <sip:182.33.174.10;lr;ftag=as0b42eef3> From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3 To: <sip:1800715080@82.160.190.100>;tag=as4133584e Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060 CSeq: 102 INVITE Server: Asterisk PBX 18.9.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces *Contact: <sip:1800715080@198.201.240.242:5080>* Content-Type: application/sdp Content-Length: 314 , ACK Reply from Provider to Us (RURI points to Kamailio ip:port now =========================================================== 2025/01/15 17:14:49.478119 182.33.174.10:5060 -> 82.160.190.100:5060 *ACK sip:1800715080@82.160.190.100:5060 *SIP/2.0 Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.2 Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK55656469;rport=5060 Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes>,sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Max-Forwards: 69 From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3 To: <sip:1800715080@82.160.190.100>;tag=as4133584e Contact: <sip:0737965510@182.33.174.39:5060> Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060 CSeq: 102 ACK Thanks -Barry

5 5

Kamailio v5.8.5 Released
by Daniel-Constantin Mierla 23 Jan '25

23 Jan '25

Hello, Kamailio SIP Server v5.8.5 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.5. For more details about version 5.8.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/01/kamailio-v5-8-5-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 Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Getting headers from xhttp/websocket
by Barry Flanagan 23 Jan '25

23 Jan '25

Hi, I need to be able to pull in a header provided by a webrtc endpoint so I can use it in my request_route. I understand that I will need to use shm/htable to make it visible but I am unable to get the xhttp header using $header(x-p-push), or any header by name. In my event_route[xhttp:request] I have the following: if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET") { .... xlog("L_INFO", "BF x-p-push header is $hdr(x-p-push) Contact is $hdr(Contact)\n"); } The log shows both x-p-push and Contact header as null. sipdump shows the headers present in the request. Am I missing something? Running Kamailio 5.8.4 Thanks -Barry

3 3

Development is open for future release of v6.1.x series
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, the branch 6.0 was created, therefore the master branch is open for adding new features, to be part of future release series v6.1.x. Any bug fix committed to master that applies to 6.0.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 6.0 or 5.8. Expect that v6.0.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v6.1.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime during the last part of 2025 or the beginning of 2026. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Branch 6.0 has been created
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, the branch 6.0 has been created, to be used for releasing v6.0.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-6.0 cd kamailio-6.0 git checkout -b 6.0 origin/6.0 Pushing commits in this branch: git push origin 6.0:6.0 Note that 6.0 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 6.0 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 6.0 will be out, respectively Kamailio v6.0.0. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Not able to get the $TV(un) inside xhttp module
by kaseenadh111＠gmail.com 22 Jan '25

22 Jan '25

Hi, I am running Kamailio 5.7.* RHEL I need to log the latency of each XHTTP request (milli or microseconds precision) so I tried using $TV(un) which always gives a 0 value. Can we access $TV(un) under XHTTP routes? event_route[xhttp:request] { $var(tv_http_req_start_time) = $TV(un); xlog("L_INFO", "$$var(tv_http_req_start_time)= $var(tv_http_req_start_time) and actual $$TV(un)= $TV(un)"); .... } Output: Jan 22 16:18:20 VM-123************ kamailio[346535]: INFO: <script>: $var(tv_http_req_start_time)= 0 and actual $TV(un)= 0 $TS - I have tried this, it gives the Unix in seconds precision but I want milliseconds precision. Thanks Kasee

1 0

Re: [sr-dev] Re: Branching 6.0 series
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, later today, in a few hours, I am going to create the branch 6.0. More details will be provided once it has been created. Cheers, Daniel On 20.01.25 08:25, Daniel-Constantin Mierla via sr-dev wrote: > Hello, > > I think we should branch it on Wednesday, Jan 22, 2025, no matter on > what stage we are with cmake support. The old-Makefiles should be kept > anyhow (as they are or in a special folder with an easy way to recover > them), because there are many bits and pieces that can be discovered > later when more users. > > Because there were no many bugs reported to the C code specific to 6.0, > we can aim releasing on January 29, 2025, with old-Makefile still to be > used for tasks that are not yet covered by cmake. > > Cheers, > Daniel > > On 15.01.25 11:16, Henning Westerholt via sr-dev wrote: >> Hello, >> >> yes, Xenofon is also waiting for a final reply from Sergey for the RPM packaging if now all is fine (was discussed as part of PR #4085). >> >> Cheers, >> >> Henning >> >>> -----Original Message----- >>> From: Victor Seva via sr-dev <sr-dev(a)lists.kamailio.org> >>> Sent: Mittwoch, 15. Januar 2025 10:59 >>> To: sr-dev(a)lists.kamailio.org >>> Cc: Victor Seva <linuxmaniac(a)torreviejawireless.org> >>> Subject: [sr-dev] Re: Branching 6.0 series >>> >>> Hi, >>> >>> please postpone. I didn't have time finalize the test of cmake builds on debian >>> packaging yet. >>> >>> On 15/1/25 7:18, Daniel-Constantin Mierla via sr-dev wrote: >>>> Hello, >>>> >>>> creating git branch 6.0 was planned for today and I wonder if it shall >>>> be done latter during the evening or it would be preferred to be >>>> postponed for two days or so, till end of the week/Friday, as I >>>> noticed yesterday some new work being done to cmake files for FreeBSD >>>> and maybe new commits are still coming. >>>> >>>> Of course, there is also the option to create the branch and then the >>>> new commits backported, but maybe postponing makes everything more >>>> convenient. >>>> >>>> In short, if anyone feels that postponing a bit creating the branch >>>> 6.0 simplifies their current work, let us know. If nobody ask for >>>> postponing, the branch will be created later towards the end of today. >>>> >>>> Cheers, >>>> Daniel >>> -- >>> ----------------------------------------------------------------- >>> | ⢀⣴⠾⠻⢶⣦⠀ Victor Seva | >>> | ⣾⠁⢠⠒⠀⣿⡁ linuxmaniac(a)torreviejawireless.org | >>> | ⢿⡄⠘⠷⠚⠋⠀PGP: 8F19 CADC D42A 42D4 5563 730C 51A0 9B18 CF5A >>> 5068 | >>> | ⠈⠳⣄⠀⠀⠀ Debian Developer | >>> ----------------------------------------------------------------- >> _______________________________________________ >> Kamailio - Development Mailing List -- sr-dev(a)lists.kamailio.org >> To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to the sender! > > -- > Daniel-Constantin Mierla (@ asipto.com) > twitter.com/miconda -- linkedin.com/in/miconda > Kamailio Consultancy, Training and Development Services -- asipto.com > > _______________________________________________ > Kamailio - Development Mailing List -- sr-dev(a)lists.kamailio.org > To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender! -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 1

Multi-line Via: rport added incorrectly
by James Browne 22 Jan '25

22 Jan '25

Hi I see that when kamailio adds rport to the Via header field of a request that has two Via values on the same line (comma-separated, of course), it adds the rport (and received) to the wrong value. I have this test kamailio.cfg for demonstration. children=1 loadmodule "sl" loadmodule "textops" loadmodule "nathelper" request_route { force_rport(); sl_send_reply(200, "OK"); } I send this OPTIONS (truncated) and get this response (truncated). The rport should be on the first Via value, not the second. OPTIONS sip:server SIP/2.0 Via: SIP/2.0/UDP 3.9.4.2:5060;branch=z9hG4bK503983089,SIP/2.0/UDP 1.3.4.1:5060;branch=z9hG4bK53805983059 From: <sip:client>;tag=LeonhardEuler To: <sip:server> SIP/2.0 200 OK Via: SIP/2.0/UDP 3.9.4.2:5060;branch=z9hG4bK503983089,SIP/2.0/UDP 1.3.4.1:5060;branch=z9hG4bK53805983059;rport=53805;received=127.0.0.1 From: <sip:client>;tag=LeonhardEuler To: <sip:server>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.49678e65 Server: kamailio (6.0.0-pre0 (x86_64/linux)) I've tested this with the latest commit (c994fb8) from kamailio. I can't think how this can be anything other than a bug. Should I log a bug report for this? James

2 4

Error when sending NOTIFYs to disconnected websocket clients
by Flavio Grossi 20 Jan '25

20 Jan '25

Hi, i have some SIP websocket clients (built using the SIP.js library) SUBSCRIBEd to other clients via kamailio v5.8.2. If one of the watchers suddenly disappears (e.g. the web page is reloaded), another connection is made and a new watcher appears in kamailio without the old one disappearing. For example: # kamctl rpc presence.watcher_list basic sip:1001@example.voismart.com { "jsonrpc": "2.0", "result": [ { "pres_uri": "sip:1001@example.com", "to_user": "1001", "watcher_user": "1000", "contact": "sip:1000@mfp0h2dhhmq2.invalid;transport=ws;alias=192.168.56.10~34166~5", "callid": "1o9b499kj6l65if0p30t", "user_agent": "sip.js/1.12.1", "expires": 1736422494, ... }, { "pres_uri": "sip:1001@example.com", "to_user": "1001", "watcher_user": "1000", "contact": "sip:1000@0j7qoabnv827.invalid;transport=ws;alias=192.168.56.10~34156~5", "callid": "8mjhrqifgm39up9t2qq3", "user_agent": "sip.js/1.12.1", "expires": 1736422492, ... } ], "id": 11753 } you can see the duplicated watcher in which the old connection is now invalid because the websocket client disappeared. In this situation, issuing a pres_refresh_watchers() call will fail to send any notify (even to the valid connection) because of this error: WARNING: <core> [core/msg_translator.c:3007]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: tm [t_msgbuilder.c:1423]: assemble_via(): via building failed ERROR: tm [t_msgbuilder.c:1614]: build_uac_req(): error while assembling Via ERROR: tm [uac.c:552]: t_uac_prepare(): Error while building message ERROR: presence [notify.c:1734]: send_notify_request(): in function tm t_request_within() ERROR: presence [notify.c:1829]: notify(): sending Notify not successful ERROR: presence [notify.c:1585]: query_db_notify(): Could not send notify for [event]=dialog ERROR: presence [presence.c:739]: pres_refresh_watchers(): sending Notify requests in which the building of the notify message fails because it cannot find a valid connection when building the via header. Is this a wanted behaviour? shouldn't it just fail the sending of the single message? Is there anything i can do to fix this? Trying to reproduce the same problem with a regular tcp client, it seems that the other notifys are correctly sent and only a timeout error is sent for the single message: ERROR: <core> [core/tcp_main.c:4622]: handle_tcpconn_ev(): connect 192.168.56.10:21994 failed which i think is the correct behaviour. Related question: is it possible to remove active watchers when the connection goes down? Thanks for the help on this.

2 5

Performance issues with KEMI
by Daniel W. Graham 20 Jan '25

20 Jan '25

Testing out KEMI functionality and running into performance issues. If I exceed 150 calls per second the network receive queue grows and Kamailio is unable to keep up with requests and they begin dropping. KEMI script for testing is just doing a stateless reply to invites. Using python3s module. I've played with Kamailio child processes and memory allocations, but there is no impact. I've also attempted some buffer / memory tweaking at the OS level, again with no impact. Increasing CPU cores and even running the test on bare metal results in the same. Example of receive queue at 150 calls per second - Netid State Recv-Q Send-Q Local Address:Port udp UNCONN 337280 0 x.x.x.x:5060 Just wondering if anyone has experienced similar issues or has an example of the performance they are seeing before I continue down this path. Thanks, - dan

5 9

Need help with serial branching using: t_load_contacts(1)
by Benoît Panizzon 20 Jan '25

20 Jan '25

Hi Gang I guess, I don't completely understand who to properly perform serial branching. I did try to follow the examples from: https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_next_con… This is, stripped down, the relevant config used. modparam("tm", "contacts_avp", "tm_contacts") modparam("tm", "contact_flows_avp", "tm_contact_flows") route[LOCATION] { lookup_to_dset("location", "$var(lookupuri)"); if (!t_load_contacts(1)) { xlog("L_ERR", "$cfg(route): ######### load_contacts failed\n"); sl_send_reply("500", "Server Internal Error - Cannot load contacts"); exit; } xlog("L_INFO", "$cfg(route): Contacts loaded: $cnt($xavp(tm_contacts))\n"); => Confirms, there is more than one contact loaded. if (!t_next_contacts()) { send_reply("480", "Not registered"); } t_set_fr(5000,1500); # Set 5 second timeout for LAB testing to quickly try the next contact. t_on_branch("BR_TO_CUST"); t_on_branch_failure("BR_TO_CUST_FAILURE"); exit; } event_route[tm:branch-failure:BR_TO_CUST_FAILURE] { xlog("BRANCH FAILED $T_reply_code to $rm message\n"); t_on_branch("BR_TO_CUST"); t_on_branch_failure("BR_TO_CUST_FAILURE"); if (t_next_contact_flow()) { xlog("BRANCH FAILED, Try Next\n"); t_relay(); } else { xlog("L_INFO", "No more flows\n"); t_reply("408", "Branch Timeout"); exit; } } What happens is: INVITE is sent to the first contact, who is replying RINGING. After 5 seconds the timeout is reached and the branch-failure route engaged. Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: ERROR: <script>: BRANCH FAILED 408 to INVITE message Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: INFO: <script>: No more flows Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: CRITICAL: tm [tm.c:1554]: ki_t_reply(): w_t_reply entered in unsupported mode To my understanding, t_next_contact_flow() should have loaded the next ds, but this does not seem to happen. What am I missing? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- 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 3

Kamailio IMS, used with IPv6?
by Valentin Christoph 16 Jan '25

16 Jan '25

Anybody experiences with Kamailio IMS modules (ims_registrar_scscf, ims_usrloc_scscf, ims_isc, ........) and the use of IPv6 addresses? Any pitfalls? DNS issues? Thanks, Christoph

2 2

Asynchttp failover
by Sergio Charrua 15 Jan '25

15 Jan '25

Hello all! just wondering if anyone knows a good way to failover with the HTTP Async module. The standard HTTP Client module (synchronous) does implement some nice functionalities for failover between URL addresses, but the HTTP Async doesn't have any failover functionality... I have implemented some functions (within kamailio.cfg) for failover upon HTTP requests, but despite it working fine, it would be better to have that functionality on the HTTP Async module, just like HTTP Client has... Any technical reason for HTTP Async module not having failover functionalities? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

2 2

Module snmpstats reporting inconsistent call success metrics on intense peaks of failed calls
by Igor Potjevlesch 15 Jan '25

15 Jan '25

Hello! We are plotting the call success rate on each of our proxies. Using the Kamailio snmpstats module, we try to obtain three counters to plot them on a graph: Total Calls Active Calls Error Calls Most of the time, everything works well, and we obtain sensible values: snmpwalk -c mycommunity -v 2c W.X.Y.Z 1.3.6.1.4.1.34352.3.1.3.1.3.2 -O n .1.3.6.1.4.1.34352.3.1.3.1.3.2.1.0 = Gauge32: 309 -> Total Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 = Gauge32: 95 -> Active Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.3.0 = Gauge32: 214 -> Error calls However, when our proxies are facing intense peaks of failed calls in a short time span, sometimes, the "Error Calls" counter returns a value higher than "Total Calls". As a result, "Active Calls" suddenly reaches a value close to the maximum of a Gauge32 (because T o t a l C a l l s − E r r o r C a l l s < 0 ). snmpwalk -c mycommunity -v 2c W.X.Y.Z 1.3.6.1.4.1.34352.3.1.3.1.3.2 -O n .1.3.6.1.4.1.34352.3.1.3.1.3.2.1.0 = Gauge32: 153 -> Total Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 = Gauge32: 4294967226 -> Active Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.3.0 = Gauge32: 223 -> Error calls As a result, our graphs make no more sense. We are currently running kamailio 5.6.4. Any idea if it's a bug or a configuration issue? Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

3 7

Branching 6.0 series
by Daniel-Constantin Mierla 15 Jan '25

15 Jan '25

Hello, I am trying to figure out what would be the best date to branch 6.0 in the git repo. To allow a bit of time after returning from the winter holidays to tune more the cmake and packaging, I would think that January 15, 2025 could be a good choice. If nobody comes with other opinions/suggestions, then we will go for it. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

kamcmd tls.reload issue
by Yuriy Nasida 14 Jan '25

14 Jan '25

Hello, I am using letsencrypt cert and key and do not want to restart kamailio every 3 months to load new ones. I know that there is: kamcmd tls.reload method but it has an error for me. error: 500 - Error while fixing TLS configuration (consult server log) I am checking the logs and see: kamailio[3865480]: INFO: tls [tls_domain.c:345]: ksr_tls_fill_missing(): TLSs<default>: tls_method=3 kamailio[3865480]: INFO: tls [tls_domain.c:357]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/certs/my_cert.crt' kamailio[3865480]: INFO: tls [tls_domain.c:364]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:371]: ksr_tls_fill_missing(): TLSs<default>: ca_path='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:378]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:382]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 kamailio[3865480]: INFO: tls [tls_domain.c:390]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:397]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/certs/private.key' kamailio[3865480]: INFO: tls [tls_domain.c:401]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 kamailio[3865480]: INFO: tls [tls_domain.c:406]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 kamailio[3865480]: INFO: tls [tls_domain.c:410]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 kamailio[3865480]: NOTICE: tls [tls_domain.c:1168]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... kamailio[3865480]: ERROR: tls [tls_domain.c:590]: load_cert(): TLSs<default>: Unable to load certificate file '/etc/kamailio/certs/my_cert.crt' kamailio[3865480]: ERROR: tls [tls_util.h:49]: tls_err_ret(): load_cert:error:03000072:digital envelope routines::decode error (sni: unknown) kamailio[3865480]: ERROR: tls [tls_util.h:49]: tls_err_ret(): load_cert:error:0A00018F:SSL routines::ee key too small (sni: unknown) Any advice ? It's interesting that there are not any errors in case I restart kamailio. I can make TLS calls without problems. deb 12.5 version: kamailio 5.7.4 (x86_64/linux)

4 5

FOSDEM VoIP Dinner
by Torrey Searle 14 Jan '25

14 Jan '25

Hello all, As in previous years, I'm organising the VoIP dinner for FOSDEM (1 Feb @ 19:00) However unlike previous years (since the restaurant got upset about splitting the bill 30 ways). I will ask everyone to order their meal and pay in advance. (and I will pay the bill to the restaurant with my card) If you are wanting to attend the dinner please fill out the following form and follow the instructions for payment. https://form.jotform.com/250012543310033 Contact details are also in the form if you want to reach out to me with any further questions. Kind regards Torrey

5 4

ims_diameter_server not freeing shared memory
by marpett365＠gmail.com 09 Jan '25

09 Jan '25

Hi, I have run in to an issue when sending diameter requests using the ims_diameter_server module. Shared memory keeps increasing and are not being freed until restart of kamailio. This leads to all shared memory is being used and no more are able to be allocated. This only happens when the request is sent synchronously. When sent asynchronously shared memory is being freed as far as I can see. have anyone ran into this issue and found a remedy or can give some pointers how to proceed? Following functions have been used for sending requests. diameter_request([peer], appid, commandcode, message) diameter_request_async([peer], appid, commandcode, message) route[CHECK_LOCATION] { diameter_request("hss.ims.test.local", "16777217", "306", "[ " + "{ \"avpCode\":260, \"vendorId\":0, \"Flags\":64, \"list\":[ { \"avpCode\":266, \"vendorId\":0, \"Flags\":64, \"int32\":10415 }, { \"avpCode\":258, \"vendorId\":0, \"Flags\":64, \"int32\":16777217 } ] }, " + "{ \"avpCode\":277, \"vendorId\":0, \"Flags\":64, \"int32\":1 }, " + "{ \"avpCode\":283, \"vendorId\":0, \"Flags\":64, \"string\":\"ims.test.local\"}, " + "{ \"avpCode\":706, \"vendorId\":10415, \"Flags\":192, \"int32\":0 }, " + "{ \"avpCode\":700, \"vendorId\":10415, \"Flags\":192, \"list\": [ { \"avpCode\":601, \"vendorId\":10415, \"Flags\":192, \"string\":\"tel:" + $fU + "\" }]}, " + "{ \"avpCode\":707, \"vendorId\":10415, \"Flags\":192, \"int32\":0 }, " + "{ \"avpCode\":703, \"vendorId\":10415, \"Flags\":192, \"int32\":14 } ]"); xlog("L_INFO", "Sent Diameter request"); } Shared memory after restart kamcmd mod.stats cdp shm Module: cdp { save_peer_applications(779): 144 make_send_pipe(126): 320 dp_add_pid(92): 216 AAAAddRequestHandler(69): 40 cb_add(183): 8 cb_add(177): 32 cdp_sessions_init(178): 8 cdp_sessions_init(173): 8 cdp_sessions_init(165): 1024 cdp_sessions_init(157): 3072 cdp_sessions_init(149): 8 cdp_trans_init(67): 8 cdp_trans_init(60): 24 add_timer(150): 24 add_timer(145): 144 new_peer(77): 16 new_peer(75): 16 new_peer(72): 16 new_peer(69): 80 new_peer(66): 96 new_peer(60): 384 peer_manager_init(84): 8 peer_manager_init(83): 8 peer_manager_init(82): 8 peer_manager_init(79): 8 peer_manager_init(76): 16 worker_init(101): 16 worker_init(94): 128 worker_init(89): 32 worker_init(87): 32 worker_init(84): 8 worker_init(82): 48 timer_cdp_init(182): 8 timer_cdp_init(179): 16 diameter_peer_init_real(137): 8 diameter_peer_init_real(131): 16 diameter_peer_init_real(122): 16 diameter_peer_init_real(115): 8 diameter_peer_init_real(108): 8 diameter_peer_init_real(101): 8 diameter_peer_init_real(94): 8 new_routing_entry(88): 64 parse_dp_config(447): 16 parse_dp_config(348): 8 parse_dp_config(340): 16 parse_dp_config(333): 40 parse_dp_config(326): 144 quote_trim_dup(96): 456 new_dp_config(52): 184 Total: 7024 } Shared memory after 10 diameter requests Module: cdp { save_peer_applications(779): 144 AAACreateAVP(137): 5072 AAATranslateMessage(500): 1320 do_receive(600): 7392 make_send_pipe(126): 320 dp_add_pid(92): 216 AAAAddRequestHandler(69): 40 cb_add(183): 8 cb_add(177): 32 cdp_sessions_init(178): 8 cdp_sessions_init(173): 8 cdp_sessions_init(165): 1024 cdp_sessions_init(157): 3072 cdp_sessions_init(149): 8 cdp_trans_init(67): 8 cdp_trans_init(60): 24 add_timer(150): 24 add_timer(145): 144 new_peer(77): 16 new_peer(75): 16 new_peer(72): 16 new_peer(69): 80 new_peer(66): 96 new_peer(60): 384 peer_manager_init(84): 8 peer_manager_init(83): 8 peer_manager_init(82): 8 peer_manager_init(79): 8 peer_manager_init(76): 16 worker_init(101): 16 worker_init(94): 128 worker_init(89): 32 worker_init(87): 32 worker_init(84): 8 worker_init(82): 48 timer_cdp_init(182): 8 timer_cdp_init(179): 16 diameter_peer_init_real(137): 8 diameter_peer_init_real(131): 16 diameter_peer_init_real(122): 16 diameter_peer_init_real(115): 8 diameter_peer_init_real(108): 8 diameter_peer_init_real(101): 8 diameter_peer_init_real(94): 8 new_routing_entry(88): 64 parse_dp_config(447): 16 parse_dp_config(348): 8 parse_dp_config(340): 16 parse_dp_config(333): 40 parse_dp_config(326): 144 quote_trim_dup(96): 456 new_dp_config(52): 184 Total: 20808 } Shared memory after 170k requests do_receive(600): 131363752 AAATranslateMessage(500): 32326656 AAACreateAVP(137): 110067152 Total: 273764840 Br Martin

2 5

evapi not processing received messages
by Sergio Charrua 09 Jan '25

09 Jan '25

Hi all! I have implemented evapi module on Kamailio 5.8.4, calling a python script. Python script is listening to a socket. It receives messages (JSON) from Kamailio, makes an HTTP request somewhere, and sends the response back to Kamailio. The Python script is working fine: it connects to Kamailio, sends HTTP requests, and forwards a response back to Kamailio: I have confirmed using TCPDump that, in fact, the data is received and forwarded back. However, despite showing messages that the Python client script is correctly connected to Kamailio, the event_route[evapi:message-received] is never executed. Probably there is some detail that I am missing, but the docs and online info for evapi are very rare, and they all show very similar configuration. And as I wrote, I have confirmed using TCPDump that the data is received from Python client script to Kamailio node. What am I missing? This is my Kamailio cfg: [loads modules] [sets modparams] modparam("evapi","bind_addr","10.20.20.1:8888") debug=2 children=16 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " disable_sctp = yes force_rport = yes rundir="/tmp" request_route { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); } exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } route(TOEVAPI); exit; } event_route[evapi:connection-new] { xlog("LINFO","new connection from [$evapi(srcaddr):$evapi(srcport)]\n"); if($evapi(srcaddr)!="10.20.0.1") { evapi_close(); exit; } } event_route[evapi:connection-closed] { xlog("LINFO","connection closed by $evapi(srcaddr):$evapi(srcport)\n"); } event_route[evapi:message-received] { xlog("LINFO","received [$evapi(msg)] from $evapi(srcaddr):$evapi(srcport)\n"); jansson_get("t-index", "$evapi(msg)", "$var(t-index)"); jansson_get("t-label", "$evapi(msg)", "$var(t-label)"); $var(evmsg) = $evapi(msg); xlog("L_INFO", "preparing to resume transaction for processing: $var(tindex) / $var(tlabel)\n"); t_continue("$var(tindex)", "$var(tlabel)", "EVAPIRESPONSE"); } route[EVAPIRESPONSE] { xlog("L_INFO", "resumed transaction for processing: $T(id_index) / $T(id_label)\n"); [do stuff] } route[TOEVAPI]{ xlog("L_INFO", "suspended transaction: $T(id_index) / $T(id_label)\n"); evapi_async_relay( [some JSON string] ); exit; } Thanks in advance! Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 6

Kamailio and Faxing
by Temilolu Fayomi 09 Jan '25

09 Jan '25

Hello, I am having some issues ensuring t38 faxes are being received reliably. I am currently using kamailio as an inbound sbc with freepbx as the endpoint. I understand that potentially stripping other codecs except for g711a/u makes inbound faxes reliable, but beyond that I am having trouble determining how to ensure reliable faxing. Is this something kamailio can handle (such as an error correction mode or pass through setting or module?) I would appreciate any insight as I have hit a wall when it comes to this issue. Thank you, Temi

3 3

Re: WS and rtpengine_manage
by Knserbrave 08 Jan '25

08 Jan '25

test

2 4

deb-archive.kamailio.org unreachable
by Floimair Florian 08 Jan '25

08 Jan '25

Hi! It seems that deb-archive.kamailio.org is currently down. It resolves to IP 76.9.245.162 on my system Can someone with access to that please have a look. Thank you FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2459749227] commend.com LG Salzburg / FN 178618z

2 2

Kamailio crash
by Mitesh Thakkar 07 Jan '25

07 Jan '25

Hi, I have recently migrated from Kamailio 5.2.5 to Kamailio 5.5.7. After a migration, the Kamailio crashed. Find the details below. Thanks in advance. Back trace: ========= Core was generated by `/usr/local/sbin/kamailio -DD -E -m 1941 -M 970 -f /etc/kamailio/kamailio.cfg --'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8e27a2be15 in OPENSSL_cleanse () from /lib/x86_64-linux-gnu/libcrypto.so.3 (gdb) bt #0 0x00007f8e27a2be15 in OPENSSL_cleanse () from /lib/x86_64-linux-gnu/libcrypto.so.3 #1 0x00007f8e278faa8a in CRYPTO_clear_free () from /lib/x86_64-linux-gnu/libcrypto.so.3 #2 0x00007f8e27838175 in BUF_MEM_free () from /lib/x86_64-linux-gnu/libcrypto.so.3 #3 0x00007f8e27bc0c98 in SSL_free () from /lib/x86_64-linux-gnu/libssl.so.3 #4 0x00007f8e27c8cd6e in tls_h_tcpconn_clean_f (c=0x7f8db0fa1e90) at tls_server.c:701 #5 0x000055f7f42cc6e1 in _tcpconn_free (c=0x7f8db0fa1e90) at core/tcp_main.c:1541 #6 0x000055f7f42e2f3c in tcpconn_put_destroy (tcpconn=0x7f8db0fa1e90) at core/tcp_main.c:3285 #7 0x000055f7f42e8759 in handle_tcp_child (tcp_c=0x7f8e27d88078, fd_i=-1) at core/tcp_main.c:3722 #8 0x000055f7f42f506f in handle_io (fm=0x7f8e27db1578, ev=1, idx=-1) at core/tcp_main.c:4539 #9 0x000055f7f42b6e49 in io_wait_loop_epoll (h=0x55f7f45ae240 <io_h>, t=5, repeat=0) at core/io_wait.h:1070 #10 0x000055f7f42f8bc3 in tcp_main_loop () at core/tcp_main.c:4824 #11 0x000055f7f4057fc2 in main_loop () at main.c:1856 #12 0x000055f7f4062670 in main (argc=10, argv=0x7ffc6d583528) at main.c:3052 OS: Ubuntu 22.04.5 LTS \n \l # kamailio --version version: kamailio 5.5.7 (x86_64/linux) 272fdd flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 272fdd compiled on 14:21:35 Oct 21 2024 with gcc 11.4.0 # dpkg -l |grep openssl ii openssl 3.0.2-0ubuntu1.18 amd64 Secure Sockets Layer toolkit - cryptographic utility Regards.

3 3

children monitoring
by Sergey Safarov 06 Jan '25

06 Jan '25

If TCP children are busy, Please correct me if I am wrong if the remote IP is not responding via TCP protocol, this leads to a timeout and this child is busy. If many remote endpoints are not reachable via TCP this can lead to the case when half or more children are by timeout operations. I think required to monitor how many children are busy. Is it possible to get such info using kamcmd utility, statistics module, or xprom module. Sergey

4 7

Crash on dialog_publish.
by satyaprakash ch 06 Jan '25

06 Jan '25

Hi, The kamailio version we are using is "5.7.4", We are facing a kamailio crash recently and it has generated the core-dump. Below is the backtrace for the coredump. Could you please assist with this crash. *Bachtrace:* #0 0x00007fa6b051748a in dialog_publish (state=0x7fa6b0519505 "terminated", ruri=0x7069734030783532, entity=0x7ffcdfb16510, peer=0x7ffcdfb16520, callid=0x7fa5b0527cb8, initiator=1, lifetime=10, localtag=0x0, remotetag=0x0, localtarget=0x7fa5b0527ce8, remotetarget=0x7ffcdfb16500, do_pubruri_localcheck=0, uuid=0x7fa5b0527d18) at dialog_publish.c:302 body = 0x0 uri = {s = 0x0, len = 0} publ = 0x0 size = 0 content_type = {s = 0x0, len = -1288318368} ruri_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = (unknown: 0), transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} __func__ = "dialog_publish" #1 0x00007fa6b0518f89 in dialog_publish_multi (state=0x7fa6b0519505 "terminated", ruris=0x7069734030783532, entity=0x7ffcdfb16510, peer=0x7ffcdfb16520, callid=0x7fa5b0527cb8, initiator=1, lifetime=10, localtag=0x0, remotetag=0x0, localtarget=0x7fa5b0527ce8, remotetarget=0x7ffcdfb16500, do_pubruri_localcheck=0, uuid=0x7fa5b0527d18) at dialog_publish.c:412 __func__ = "dialog_publish_multi" #2 0x00007fa6b0506944 in __dialog_sendpublish (dlg=0x7fa5b046a850, type=64, _params=0x7fa6b0437620 <params>) at pua_dialoginfo.c:402 tag = {s = 0x0, len = 0} uri = { s = 0x7fa5b0527d41 "sip:+1248XXXXXXX@10.XXX.X.Xm6ka0542f9v0lmjjc4npkamailio954739148rdb2847sip :XXXXXX.com:5060sip:521XXXx0;tgrp=51;trunk-context=198.X.XXX.XXX@10.XXX.XX.XX:5060padi-6761ad46-154e-1560padi\300\300\300\300", len = 29} identity_local = { s = 0x7fa5b0527d28 "sip:413XXXx0@10.XXX.X.XXsip :+1248XXXXXXXX@10.XXX.X.XXm6ka0542f9v0lmjjc4npkamailio954739148rdb2847sip :XXXXXX.com:5060sip:415XXXx0;tgrp=51;trunk-context=198.X.XXX.XXX@10.XXX.XX.XX:5060padi-6761ad46-154e-"..., len = 25} target = {s = 0x0, len = 0} dlginfo = 0x7fa5b0527c90 request = 0x7fa6b3361b58 __func__ = "__dialog_sendpublish" Thanks.

1 2

record_route() adding wrong sips URI when transport is missing (defaulting to UDP)
by Benoit Panizzon 06 Jan '25

06 Jan '25

Hi Gang A quick search via google did not reveal any useful hits. Kamailio 5.7.6 CPE (TLS) => SBC (UDP) => Registrar (UDP) => Core CPE uses a sips contact URI to register, no transport attribute! sips:315996608@172.29.255.11:5061 SBC uses Path functionality. Situation: Call to CPE I see the registrar adding this Record-Route: Record-Route: <sips:registrar-ip;lr;ftag=6159862378> The SBC is adding a r2=on Record-Route to indicate the change between UDP and TLS. The Route-Set received by the CPE then is: INVITE sips:315996608@172.29.255.11:5061 SIP/2.0 Record-Route: <sips:sbc-ip:5061;transport=tls;r2=on;lr;ftag=6159862378> Record-Route: <sips:sbc-ip;r2=on;lr;ftag=6159862378> Record-Route: <sips:registrar-ip;lr;ftag=6159862378> Record-Route: <sip:core-ip;lr;ftag=6159862378> 180 Ringing req 100rel is making it back as the Via header look right. But when the remote side is sending a PRACK, is has to follow the Routeset which if I am not mistaking, is not making it back because because the CORE does not know how to send this via TLS to the Registrar. Core logging: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) IMHO already the 2nd Record-Route added by the Registrar should not contain a sips: URI but a sip: URI (reflecting the transport attribute not being present, defaulting to UDP) Is this a known issue? 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 ______________________________________________________

3 4

Guidance Required for Kamailio Load Balancer Behavior in Attended Transfer Scenarios
by Kotappa Gandudi 03 Jan '25

03 Jan '25

Dear Kamailio Team, I hope this email finds you well. I am currently working on a project where Kamailio is being utilized as a load balancer to distribute calls across multiple FreeSWITCH instances. I am using the round-robin technique for load distribution, and it is working perfectly for regular call scenarios. However, I am encountering an issue when handling attended transfer calls. Specifically, for a normal call, Kamailio assigns one FreeSWITCH instance, but during the attended transfer call, it assigns another FreeSWITCH instance. This behavior disrupts the continuity of the call handling. My requirement is for Kamailio to use the same FreeSWITCH instance for the entire call, including any attended transfers. I believe maintaining the same FreeSWITCH instance for the entire call flow would ensure consistency and avoid potential issues. Could you please guide me on how to configure Kamailio to achieve this behavior? I would greatly appreciate any suggestions, configurations, or modules that can help resolve this issue. Thank you for your time and assistance. Looking forward to your response. Best regards, Kotappa Y Gandudi

4 4

how to call loose_route() after adding route header without using msg_apply_changes?
by Benoît Panizzon 31 Dec '24

31 Dec '24

Hi List To handle CPE which can only handle a limited number of route header (PacketCable SIP specification), I need to store and restore the route header (limited topology hiding only affecting Route Header) to present the CPE only one Route or RR header. I pass the ID of the stored route-set via RR param. So when I get a message back from the CPE is: * Take ID from RR param and pull stored route set from cache. * remove_hf("Route"); * insert_hf("Route: $avp(stored-routeset)\r\n"); * msg_apply_changes(); * loose_route(); I found loose_route() only to be able to correctly determine the next hop, if I first did apply the changes. But after adding rtpengine to the mix, I got back into the known situation that calling rtpengine (on the branch route) after calling msg_apply_changes() results in duplicate SDP. Is there a way to get loose_route() to operate on freshly added header without calling msg_apply_changes? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- 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 ______________________________________________________

8 28

Async HTTP request is sent but response is incorrectly processed
by Sergio Charrua 30 Dec '24

30 Dec '24

Hi all! still struggling with Async HTTP Client ... My current configuration is very simple: #!define HTTP_API_ROUTING_ENDPOINT_MAIN " http://MY_HTTP_REST_API:18085/Endpoint" include_file "listen.cfg" loadmodule "pv" loadmodule "tm" loadmodule "http_async_client" loadmodule "xlog" loadmodule "tmx" loadmodule "sl" loadmodule "textops" loadmodule "cfgutils" loadmodule "siputils" loadmodule "rr" loadmodule "ctl" loadmodule "jsonrpcs.so" loadmodule "jansson.so" modparam("pv", "shvset", "http_async=i:8"); modparam("http_async_client", "workers", 1); modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "curl_verbose", 1) # for debug purpose #modparam("tm", "fr_timer", 750) #modparam("tm", "fr_inv_timer", 20000) #modparam("tm", "remap_503_500", 0) modparam("tm", "failure_reply_mode", 3) modparam("tm", "max_inv_lifetime",5000) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") debug=2 children=16 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " disable_sctp = yes force_rport = yes rundir="/tmp" request_route { $avp(REQUEST_URL) = HTTP_API_ROUTING_ENDPOINT_MAIN; if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); } exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } jansson_set("string","from" , $hdr(From) , "$var(req)"); jansson_set("string","to" , $hdr(To) , "$var(req)"); jansson_set("string","r-uri" , $ru , "$var(req)"); jansson_set("string","contact" , $hdr(Contact) , "$var(req)"); jansson_set("string","call-id" , $ci , "$var(req)"); xinfo("MAIN-ROUTE - Setting From=$hdr(From) To=$hdr(To) Contact=$hdr(Contact) ci=$ci \n"); if ( is_present_hf("Identity") ) jansson_set("string","identity" , $hdr(Identity) , "$var(req)"); if ( is_present_hf("P-Identity-Bypass") ) jansson_set("string","p-identity-bypass" , $hdr(P-Identity-Bypass) , "$var(req)"); if ( is_present_hf("P-Asserted-Identity") ) jansson_set("string","p-asserted-identity" , $hdr(P-Asserted-Identity) , "$var(req)"); if ( is_present_hf("P-STSH-UC") ) jansson_set("string","p-stsh-uc" , $hdr(P-STSH-UC) , "$var(req)"); jansson_set("string", "request-time" , $avp(requestTime), "$var(req)"); ## Create transaction before http_async_query; if (is_method("INVITE")){ t_newtran(); xinfo("MAIN-ROUTE - Sending request [$var(req)]\n"); route(HTTP_ASYNC_REQUEST); } exit; } route[HTTP_ASYNC_REQUEST] { xinfo("HTTP_ASYNC_REQUEST - Sending request [$var(req)]\n"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = 1500; # 1 second timeout $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; $http_req(body) = $var(req); if ( !http_async_query($avp(REQUEST_URL), "HTTP_REPLY") ) { xinfo("HTTP_ASYNC_REQUEST - http_async_query $ci FAILED. Return is [$rc]\n"); }else{ send_reply("503", "Server Error"); exit; } } route[HTTP_REPLY] { xinfo("HTTP_REPLY - Xphere Response Code $http_rs - $ci\n"); if ( $http_ok && $http_rs==200) { xinfo("HTTP_REPLY - $ci reply ok\n"); jansson_xdecode($http_rb, "json"); xinfo("HTTP_REPLY - HTTP JSON Response: $http_rb \n"); if ( $xavp(json=>sip-response-code) == 300 ){ remove_hf("Contact"); append_to_reply("Contact: $xavp(json=>contact)\r\n"); ## "touch" the ruri to add the contact header for 3xx: $ru = $ru; send_reply("300", "Multiple Choice"); exit; }else{ xinfo("HTTP_REPLY - $ci Error Replied $xavp(json=>sip-response-code) - $xavp(json=>sip-response-text) \n"); send_reply("$xavp(json=>sip-response-code)","$xavp(json=>sip-response-text)"); exit; } } else { xinfo("HTTP_REPLY - HTTP Response Code $ci - $http_rs and error $http_err \n"); if (!$http_ok){ sl_reply("503", "No Connection to HTTP"); exit; }else if ($http_rs>=500){ sl_reply("503", "HTTP Processing Error"); exit; } else{ sl_reply("503", "Cannot Connect to HTTP"); exit; } } exit; } The configuration is based on Ben Kaufman's kamailio_http_async <https://github.com/whosgonna/kamailio_http_async> example, but I have extended it a bit. Currently all the requests are being sent to the HTTP REST API. I can't find queue/buffering issues... What I am experiencing is that , using SNGREP, I get this flow: 10.240.12.209:5060 10.242.17.146:5060 ----------*---------- ----------*---------- 17:33:46.689852 x INVITE (SDP) x +0.000530 x --------------------------> x 17:33:46.690382 x 100 trying -- your call is x +1.499822 x <-------------------------- x 17:33:48.190204 x 503 No Connection to HTTP x +0.000029 x <-------------------------- x 17:33:48.190233 x 500 I'm terribly sorry, se x +0.002130 x <-------------------------- x 17:33:48.192363 x ACK x +0.485836 x --------------------------> x 17:33:48.678199 x 500 I'm terribly sorry, se x +0.004881 x <<<------------------------ x 17:33:48.683080 x ACK x +0.973295 x ------------------------>>> x 17:33:49.656375 x 500 I'm terribly sorry, se x +0.004105 x <<<------------------------ x 17:33:49.660480 x ACK x +1.995420 x ------------------------>>> x 17:33:51.655900 x 500 I'm terribly sorry, se x +0.005361 x <<<-------------------------x 17:33:51.661261 x ACK x x ------------------------>>> x I suspected issues with the HTTP REST service, but when using tcpdump, I can find the Request from Kamailio and a reply back from the REST API service, so doesn't seem to be a problem on the HTTP REST service So my questions are: - why is Kamailio sending a "SIP 500 I'm terribly sorry..." error after a 503 No Connection to HTTP is sent and an "exit" command is executed? - how can I get rid of these messages? - Why Kamailio keeps sending responses back to SBC after the ACK when it is set to just "exit"? - Why is Kamailio not processing the response from HTTP REST API when it is correctly received? Logs are also weird: - I find a lot of these warnings: "Dec 29 17:38:34 ire-nprod-sip1 /usr/local/sbin/kamailio[590864]: WARNING: http_async_client [http_async_client_mod.c:529]: ah_get_status(): an async variable was read after http error, use $http_ok to check the request's status" and "Dec 29 17:38:34 ire-nprod-sip1 /usr/local/sbin/kamailio[590864]: ERROR: http_async_client [http_multi.c:621]: check_multi_info(): handle 0x7f375f786b30 returned error 7:" (I am still not using EVAPI module, that will be my next step, after the holiday season) (I suspect, however, that the REST API is not being able to handle the load in this test environment (between 400 to 600 CAPS), but that is another matter.) In attach, TCPDUMP, if anyone wishes to analyse. Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 5

http sync / async final tests
by Alexis Fidalgo 26 Dec '24

26 Dec '24

Just to share, and thanks for your valuable help. My conclusion after tests (on my scenario) is http backend: is a golang http server with a 250ms delay to answer, and answering back a json, running in 10 replicas inside a K8 with 4 nodes. test (from kamailio perspective): receive INVITE send 100 call the http, get the result, get a key from the result and set it as the value of Contact header send 302 receive ACK (discard on sync scenario because it’s stateless, use it in async to end transaction) Kamailio host: 1vcpu kamailio 3.8.4 bookworm running inside containerd (with podman) 16GB RAM sipp: run from the K8 nodes hosts (directly from OS) (everything is LAN connected on 2 different physical servers, hypervisor is VMWare i guess 7.0.3) using http_client (sync with stateless): 8 children -M8 -m64 no more than 60 calls/sec (15cps on each sipp instance) until sipp start to retransmit the INVITES using async_http_client: 8 children -M8 -m2048 it’s been ~36 hours now at 250cps each sipp instance with 0 errors 0 retransmission and the kamailio host is userlab@sblb1:~$ vmstat 1 procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 4 0 0 10285460 364684 5444240 0 0 0 22 36 68 0 0 100 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5284 6822 27 14 58 0 0 4 0 0 10285460 364684 5444240 0 0 0 0 5306 6752 29 15 57 0 0 3 0 0 10285460 364684 5444240 0 0 0 0 5304 6823 26 16 57 0 0 2 0 0 10285460 364684 5444240 0 0 0 12 5287 6761 26 16 57 0 0 2 0 0 10285460 364684 5444240 0 0 0 0 5337 6838 27 14 58 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5101 6724 25 17 59 0 0 3 0 0 10285460 364684 5444240 0 0 0 0 5066 6650 26 14 60 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5306 6829 23 17 59 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5219 6725 23 19 58 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5274 6770 26 13 61 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5295 6756 26 18 56 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5235 6786 22 16 62 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5315 6833 21 20 59 0 0 at least for my scenario, it worths to pay the price of the memory tradeoff. again, Ben, Alex, Sergio, all, thanks for your help. regards

1 0

help on how to get ACK
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello all, moving just a bit aside of the http and async_http. After all the real useful and interesting thread on that topic what helped me, im facing a problem i cant deal with and need a hint at least. Scenario INVITE -> Kamailio on request_route ... if(is_method("INVITE")){ t_newtran(); http_async_query("http://nuc:8080", "HTTP_REPLY"); } … Kamailio -> 100 - Trying then route[HTTP_REPLY] { if ($http_ok) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); t_reply(302,"Redirect"); } else { xlog("L_INFO", "route[HTTP_REPLY]: error $http_err)\n"); } } Kamailio -> 302 Redirect ACK -> Kamailio This last ACK, how can i read it and use it to terminate the transaction? because Kamailio keeps transmitting the 302 message 3 more times until the transaction is finished by a timer 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [timer.c:642]: wait_handler(): finished transaction: 0x7f4c3fcd35a0 (p:0x7f4c3fad85d0/n:0x7f4c3fad85d0) 42(44) DEBUG: tm [h_table.c:133]: free_cell_helper(): freeing transaction 0x7f4c3fcd35a0 from timer.c:651 in request_route i have if(is_method("ACK")){ if(!t_check_trans()){ t_release(); } exit; }

4 26

append_branch
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello all, how can avoid to append_branch do this call to call SIP/2.0 302 Redirect Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-65665-5-0 From: sipp <sip:sipp@192.168.86.250:5060>;tag=65665SIPpTag005 To: 123456 <sip:123456@192.168.86.128:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-deb8da2e Call-ID: 5-65665(a)192.168.86.250 CSeq: 1 INVITE Contact: <sip:4093823670@otherdomain.com>, <sip:8888576523@otherdomain.com>, <sip:9719533197@otherdomain.com>, <sip:5302579529@otherdomain.com>, <sip:4972010218@otherdomain.com> Server: kamailio (5.8.4 (x86_64/linux)) Content-Length: 0 What i mean with “avoid”, call to call, contact should have only 1 branch, no a stack of previous. thanks

2 1

Transactions help
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello, after the discussion regarding http and http_async, I started some labs to make a comparison and see what’s the trade off and the associated cost. Main problem (to me) is im not clear on how transactions behave. Im running some fast tests now before I start to deploy the final scenarios and 1. A new transaction is created before the http_async call (as the module example shows for http_async_query function 2. The http_async_query call is executed, receives a response and the HTTP_REPLY route for http_async_query is executed. 3. As we need to redirect the call, im adding Contact information with append_to_reply and then calling t_reply(302,”Redirect) Up to here INVITE is received, a 100 is answered and then the 302 with the expected Contact information (collected and built from the http_async_query response), problem is that the 302 is retransmitted 3 more times. So, im not even sure that im doing the correct actions to process the call this way. Is there a document/book/reference I can read to understand transactions better? By now I can not handle the call to be processed correctly (without the retransmissions) and finish the transaction the correct way. Any help, link, guide, will be appreciated. Regards.

6 17

Kamailio not receiving packets on high CPS
by Sergio Charrua 23 Dec '24

23 Dec '24

Hello! for this ST/SH project, we are using Kamailio 5.8.4 in stateless mode, making HTTP requests to a REST API (Java Spring) that will reply with a JSON object, and then Kamailio replaces the Contact header and sends a SIP 300 Multiple Choice reply to the SBC that sent the initial INVITE. Kamailio is running on a VM with 4vCPUs and 4GB RAM and 2 NICs. On NIC ens224 there is a Virtual IP managed by Keepalived. Kamailio is listening on port 5060 on ens224 and set with 32 children process (tried with 8, 16, 24 and also 64, but with this value it was a lot worst). We are doing load performance tests on our PreProd environment. With the SBC we are sending 450 to 600 CAPS to Kamailio, and what we noticed is that above 450 CAPS, and after less than 1 minute, Kamailio only replies SIP 100 to some INVITEs. We also could figure out that Kamailio is not receiving all the INVITES, despite having proved that ALL invites are sent to the server. It seems to me that, for some reason, the OS is somehow not able to deliver *all* SIP packets to Kamailio, because: - sngrep does capture all the SIP packets and shows the flows - though the flow shows SIP invite coming from SBC to Kamailio Server, there is no SIP 100 replied back to SBC, even though this is the 1st thing kamailio does, so, for me, Kamailio did not received the SIP Invite - picking a call with reinvites, which after some seconds, SBC cancels, if I copy the Call-ID value and grep on kamailio's logs, I just do not find any content! I have read some "network tuning" articles, including Linux Tune Network Stack (Buffers Size) To Increase Networking Performance - nixCraft <https://www.cyberciti.biz/faq/linux-tcp-tuning/> and Tuning Kamailio for high throughput and performance | Evariste Systems Blog <https://blog.evaristesys.com/2016/02/15/tuning-kamailio-for-high-throughput…> and when running the following command I get: $ ss -4 -n -l | grep 5060 udp UNCONN 0 0 10.242.17.125:5060 0.0.0.0:* udp UNCONN 324224 0 10.242.17.146:5060 0.0.0.0:* According to the article, the 3rd column should be as near 0 as possible, ideally 0. While I am waiting for some sysadmin with root permissions to be available and modify some of the network parameters on this VM, I wonder if anyone has some tips to share on how to solve this behaviour. I'm not sure, either, what parameters to change or if they should be changed! The goal is to have a minimum 750 CAPS. Lastly, some parts of the code: Kamailio.cfg: > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=1 > log_stderror=no > rundir="/tmp" > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > # configure the prefix for all log messages > log_prefix_mode = 1 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=32 # ------------- ALREADY TRIED WITH 8,16,24... no results..... > /* uncomment the next line to disable TCP (default on) */ > disable_tcp=no > /* 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" > ####### 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 */ ####### Modules Section ######## /* set paths to location of modules */ loadmodule "db_mysql.so" loadmodule "db_cluster.so" loadmodule "http_client.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "jansson.so" loadmodule "snmpstats.so" loadmodule "file_out.so" loadmodule "ctl.so" loadmodule "permissions.so" loadmodule "xhttp.so" loadmodule "xhttp_rpc.so" ####### Other Args and Env Vars ######### /* listen addresses */ include_file "listen.cfg" include_file "db_conn.cfg" # ----------------- setting module-specific parameters -------------- # --- DB Cluster params --- # minimum requirement is to have DBURL1 defined #!ifdef DBURL1 modparam("db_cluster", "connection" , DBURL1) #!endif #!ifdef DBURL2 modparam("db_cluster", "connection" , DBURL2) #!endif #!ifdef DBURL3 modparam("db_cluster", "connection" , DBURL3) #!endif #!ifdef DBURL4 modparam("db_cluster", "connection" , DBURL4) #!endif #!ifdef DBURL5 modparam("db_cluster", "connection" , DBURL5) #!endif modparam("db_cluster", "cluster", DBCLUSTER) # --- Permissions params --- modparam("permissions", "db_url", "cluster://k1") modparam("permissions", "db_mode", 1) #modparam("permissions", "reload_delta", 30) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") modparam("jsonrpcs", "transport", 0) # ----- ctl params ----- /* set the path to RPC unix socket control file */ modparam("ctl", "binrpc", "unix:/tmp/kamailio_ctl") # ----- http_async_client params ----- modparam("http_client", "query_result", 0) modparam("http_client", "keep_connections", 0) modparam("http_client", "connection_timeout",2) modparam("http_client", "timeout_mode",2) modparam("http_client", "config_file", "/usr/local/etc/kamailio/http_client.cfg") # ---- SNMP Stats params ---- modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "sipEntityType", "redirectServer") modparam("snmpstats", "sipEntityType", "other") modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "MsgQueueMinorThreshold", 20) modparam("snmpstats", "MsgQueueMajorThreshold", 100) modparam("snmpstats", "dlg_minor_threshold", 20) modparam("snmpstats", "dlg_major_threshold", 100) modparam("snmpstats", "snmpCommunity", "kamailio") # ---- File_Out params ---- modparam("file_out", "base_folder", "/data/sabire002/log/kamailio/") modparam("file_out", "file", "name=stsh_requests;interval=1440;extension=.log") modparam("file_out", "file", "name=cdr;interval=1440;extension=.log") modparam("file_out", "file", "name=http;interval=1440;extension=.log") ####### Routing Logic ######## include_file "includes/reqinit.cfg" include_file "includes/handle_options.cfg" include_file "includes/handle_cancel.cfg" include_file "includes/handle_stir_shaken_stateless.cfg" include_file "includes/handle_http_rpc.cfg" request_route { $avp(START_TIME)=$utimef(%Y-%m-%d %H:%M:%S); $avp(GROUPID) = allow_address_group($si, $sp); if ( $avp(GROUPID) == 100 || !allow_address_group($si, $sp) ) { xlog("L_INFO", "INIT - $si:$sp is not in the allowed ACL Group ID !\n"); #sl_reply("401", "Address not allowed"); exit; }; if (is_method("ACK") ) { #&& t_check_trans() ){ exit; } if (is_method("INVITE")) { file_out("cdr","$rm|$ft|$tt|$ci|$rs|$rr|$Ts|$avp(START_TIME)|$fU|$fd|$si|$tU|$rU|$rd|$utimef(%Y-%m-%d %H:%M:%S)"); send_reply("100","Trying"); } route(HANDLE_OPTIONS); route(REQINIT); xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","=================================================== \n"); xlog("L_INFO","New SIP message $rm with call-ID $ci received $pr request $rm $ou source $si:$sp from $fu to $tu \n"); xlog("L_INFO","=================================================== \n"); route(HANDLE_STIRSHAKEN); route(HANDLE_CANCEL); if (method == "INVITE"){ route(RELAY); } } route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","RELAY - send reply \n"); file_out("cdr","RELAY|$ft|$tt|$ci|$rs|$rr|$Ts|$avp(START_TIME)|$fU|$fd|$si|$tU|$rU|$rd|$utimef(%Y-%m-%d %H:%M:%S)"); send_reply("300", "Multiple Choices"); exit; } As for the HANDLE_STIRSHAKEN route, where the main process is: route[HANDLE_STIRSHAKEN] { if (!is_method("INVITE")) { return; } $xavp(requestTime) = $utimef(%s); $var(post) = $null; // resets and makes sure the $var(post) variable is null before usage [.... get some header values ...] $var(res) = http_connect("api1", "/stsh", "application/json", "$var(post)", "$var(result)"); jansson_xdecode($var(result), "json"); if ( $xavp(json=>sip-response-code) == 300 ) { remove_hf("Contact"); append_to_reply("Contact: $xavp(json=>contact)\r\n"); } else { # add error message to reply sl_reply("$xavp(json=>sip-response-code)","$xavp(json=>sip-response-text)"); exit; } } Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

8 73

How to use @via[idx] ?
by Benoit Panizzon 23 Dec '24

23 Dec '24

Hi https://www.kamailio.org/wikidocs/cookbooks/5.7.x/pseudovariables/#selname-… https://www.kamailio.org/wikidocs/cookbooks/5.7.x/selects/#viaihost I'm trying to list all via hosts: $var(dooh) = @via[1].host; xlog("L_INFO","******** VIA HOST 1: $var(dooh) \n"); This above works as expected I get the 2nd entry. $var(i) = 0; while ($var(i) < $hflc(Via)) { $var(dooh) = @via[$var(i)].host; xlog("L_INFO","******** VIA HOST: $var(dooh) \n"); $var(i) = $var(i) + 1; } This fails with syntax error. How do I specify the index of the via header I want to extract the host from? I also attempted $sel(via[$var(i)].host) or via[$var(i)] or $sel(hfl(via[$var(i)])).host but I found no variant not throwing an error. PS: Is there maybe another nice function to figure out if we have already seen (and how many times we have) this same message? 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 ______________________________________________________

3 5

KAMI python3 and python3s not working / reloading
by trle.fuad＠gmail.com 20 Dec '24

20 Dec '24

Hello, I've started testing/learning KEMI and decide to use python3(s). I'm using default config with kamailio-basic-kemi-python3s.py (just deleted a couple of routes for simplicity). First thing I noticed is that reloading is not working. I've changed simple log line and did kamcmd app_python3s.reload. I'm compiling from source (inside Docker container, Debian bookworm as base), and have noticed it stopped working after thees commits for python3s: 62b4ee4a0d0b62b35c8bdf67e5daf9cbe9a28499 app_python3s: refactor GIL and thread state handling c81b2b3383c5900aba83672da024c812d8e6d89d app_python3s: initial support for free-threading Python And for python3 0ffe157bc13e7759ae1cee63a584fad4ac9eb38f app_python3: refactor GIL and thread state handling Am I missing some new parameter for module or compile flag? What I've noticed it's not only that it not reload, it also does not go through complete config file. While reloading, I see marking for reload. But nothing after that: 22(774) DEBUG: ctl [../../core/io_wait.h:368]: io_watch_add(): processing io_watch_add(0x748d1dfb83c0, 11, 3, 0x5f0adde454d0) - fd_no=1 22(774) DEBUG: ctl [io_listener.c:438]: handle_new_connect(): new connection (1) on /var/run/kamailio//kamailio_ctl 22(774) DEBUG: ctl [io_listener.c:499]: handle_stream_read(): bytes read: 29 22(774) INFO: app_python3s [apy3s_kemi.c:868]: app_python3s_rpc_reload(): marking for reload Python script file: /etc/kamailio/kamailio-basic-kemi-python3s.py (0 => 1) 22(774) DEBUG: ctl [io_listener.c:520]: handle_stream_read(): bytes processed: 29 22(774) DEBUG: ctl [io_listener.c:496]: handle_stream_read(): handle_stream read: eof on /var/run/kamailio//kamailio_ctl 22(774) DEBUG: ctl [../../core/io_wait.h:599]: io_watch_del(): DBG: io_watch_del (0x748d1dfb83c0, 11, -1, 0x10) fd_no=2 called

1 0

nethelper module - need clarification on how to handle uri alias on location contacts.
by Benoît Panizzon 20 Dec '24

20 Dec '24

Hi Gang When a CPE behind nat registers, I have the alias added to the AOR contact. It has format ip~port~transport Let's say: bob@1.2.3.4:5060;alias=5.6.7.8~7777~1 On an invite to that CPE behind nat, the contact is retrieved from the location record and I assumed I could call handle_ruri_alias in the branch route to that CPE and this would sent the call to that ip port via transport in the alias. I observe that the port is not taken into account. The INVITE to bob is sent to ip 5.6.7.8 but to port 5060 instead of 7777 In the example config, handle_ruri_alias() is only used for in dialog requests (has_totag). Have I misunderstood how CPE behind NAT should be registered and that registration resolved? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- 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 ______________________________________________________

1 0

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