- sr-users - kamailio.org

limit the length of Via branch parameter
by Patrick Karton 15 Mar '22

15 Mar '22

Hello, i would like to know if there is a way or option to limit the length of the value of branch parameter in Via header ? Thanks.

2 3

Table : address - address_reload (exported MI function) - expect to call this function in kamailio.cfg to reload in cache the address
by Youssef Boujraf 14 Mar '22

14 Mar '22

Dear, I use the address table to ENABLE UAC doesn't registererd in subscriber table just for emergency call then after the BYE to I remove it from the address table. I 'd like to purge the UAC from address table BUT NEED to RELOAD in CACHE the address table. To use the "kamctl address relaod" works BUT I want to reload the address table from my Kamailio.cfg (route). I found the function "address_reload" ( https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528769) but I have an error when trying to start kamailio. Best regards, -- Youssef BOUJRAF | Managing Director Direct Line: +32 (0)2 880 33 30 Mobile: +32 (0)477 94 86 89 Dubai Mobile : +971 (0)55 9928871 Fax: +32 (0)2 318 82 21 Skype: y_boujraf Email: yboujraf(a)by-research.be Web : www.by-systems.be/ BY-SYSTEMS sprl| Avenue Charles Plisnier 13 | B-1070 Anderlecht | Belgium | TVA: BE 0819.832.617______________________________________________________________________________ This communication, issued by BY-SYSTEMS SPRL, is confidential and we do not waive confidentiality by mistransmission. If you have received it by mistake please notify the sender immediately by reply and then delete it permanently from your system without copying it, using it for any purposes or disclosing its contents to any other person. Thank you for your cooperation. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of BY-SYSTEMS SPRL. Emails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, lost or destroyed, or contain viruses. Anyone who communicates with us by email is taken to accept these risks.

2 1

WebSocket ping is not being sent
by Xuo Guoto 14 Mar '22

14 Mar '22

Hello list, I have kamailio 5.5.3 (x86_64/linux) configured as an SBC in front of asterisk to front end webrtc clients. Browser clients running JsSIP 3.7.0 connect to kamailio via webrtc and kamailio sends the traffic back to asterisk using SIP. rtpengine is configured to relay the RTP packets. I need to send websocket pings every 10 seconds to the browser clients. I have the following config in kamailio: modparam("websocket", "keepalive_mechanism", 1) modparam("websocket", "keepalive_timeout", 10) modparam("websocket", "keepalive_interval", 2) modparam("websocket", "keepalive_processes", 2) modparam("websocket", "verbose_list", 1) If my understanding of the documentation is correct, I do not need to do any additional configuration. But looking at the websocket traffic, I do not see any ping packets in websocket. I am also using sipdump module to capture all packets sent and received by kamailio and the ping packets are not visible there either. What could be wrong here? Do I need to do any additional config? Thanks for every help! X.

3 5

Any clever ways to strip down a user portion of an RURI?
by Brooks Bridges 12 Mar '22

12 Mar '22

Have a need to strip out various parameters from the user portion of the RURI and/or the To (npdi, etc). Does anyone have any clever methods for doing so that don't involve just brute forcing through the individual parameters? Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

3 4

Interconnect with Zoom
by Matthias Urlichs 11 Mar '22

11 Mar '22

Hi, has anybody managed to connect to Zoom with Kamailio, ideally using rtpengine? I'm getting error 500 when I call them … -- -- regards, -- -- Matthias Urlichs

3 4

Kamailio + RADCLI
by Marrold 11 Mar '22

11 Mar '22

Hi, We're currently testing Kamailio 5.5.3 on Debian 11 compiled with RADCLI and we're experiencing an issue with radius authentication: Feb 23 08:54:22 redacted /sbin/kamailio[817172]: WARNING: <script>: RADIUS auth failed for redacted (IP:redacted:5060) Feb 23 08:54:22 redacted /sbin/kamailio[817173]: radcli: rc_avpair_new: rc_avpair_new: no attribute 1/1 in dictionary Feb 23 08:54:22 redacted2 /sbin/kamailio[817173]: ERROR: auth_radius [sterman.c:204]: add_cisco_vsa(): unable to add Cisco-AVPair attribute I don't believe this is directly a Kamailio issue but enabling debug logging hasn't revealed any clues ( The log lines remain the same without additional info ) This was working previously on 5.5.X on Debian 9 Has anyone experienced this before or have any suggestions on where to start looking? Thanks

2 2

Kamailio 5.5 TLS critical log in Debian 11
by mayamatakeshi 11 Mar '22

11 Mar '22

I'm testing with this kamailio version: $ kamcmd version kamailio 5.5.4 (x86_64/linux) 54c9df on: $ cat /etc/issue Debian GNU/Linux 11 \n \l with: $ openssl version OpenSSL 1.1.1k 25 Mar 2021 Whenever I start kamailio I get 3199 log lines like this: 2022-03-10T10:04:39.700676+09:00 lab002201-flip-server /usr/local/src/git/kamailio-5.5/src/kamailio[261703]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f8eb57f7848), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_free(323) - ignoring I searched the issues at github and found: https://github.com/kamailio/kamailio/issues/2560 https://github.com/kamailio/kamailio/issues/2912 So I tried to start kamailio with --atexit=no but those logs remain. With and without using --atexit=no there is no crash and all my tests are OK. But should I be worried to put this in production?

3 3

add a=fmtp: if it does not exist in sdp
by Krzysztof Drewicz 11 Mar '22

11 Mar '22

Hello i try to fix one endpoint, it does neet do have a a=rtpmap:116 telephone-event/8000 a=fmtp:116 0-15 sometimes i got: a=rtpmap:115 telephone-event/8000 and sometimes: a=rtpmap:114 telephone-event/8000 a=fmtp:114 0-15 so - should i use txtops and search for fmtp, and use search_append_body ? ifsearch_body(re) does not find 'a=fmtp:95' (95..127 - dynamic payload here), or am I reinventing the wheel again ? Any hints? -- Krzysztof Drewicz Senior Infrastructure Administrator CLUDO | ul. Grochowska 306/308, 03-840 Warszawa t+48221223977 kdrewicz(a)cludo.pl | www.cludo.pl

3 3

Diversion Header & Header management - $hdrc
by Brandon Armstead 10 Mar '22

10 Mar '22

Thanks for your time in advance! When dealing with multiple headers in different formats, i.e. Diversion: <sip:foo>, <sip:bar> $hdrc(Diversion) returns 1 header instead of 2. Is this expected behavior? What is the best way to manage all scenarios of possible multiple header values? It seems currently there is no easy way to iterate over ALL diversion headers, including scenarios like the following below: Diversion: <sip:abc> or Diversion: <sip:abc>, <sip:def> or Diversion: <sip:abc> Diversion: <sip:def>, <sip:ghi> etc. - Brandon

2 1

Kamailio 5.4.7 crash
by Igor Olhovskiy 10 Mar '22

10 Mar '22

Hello! My Kamailio 5.4.7 on CentOS 7 [root@pbxphone-0866-fe-1 kamailio_crash]# kamailio -V version: kamailio 5.4.7 (x86_64/linux) a19fe1 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_BLACKLIST, HAVE_RESOLV_RES 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: a19fe1 compiled on 20:40:47 Nov 22 2021 with gcc 4.8.5 [root@pbxphone-0866-fe-1 kamailio_crash]# uname -a Linux pbxphone-0866-fe-1.company.domain 3.10.0-1160.15.2.el7.x86_64 #1 SMP Wed Feb 3 15:06:38 UTC 2021 x86_64 x86_64 x86_64 GNU/Linux crashed with following backtraces (3 core files) [root@pbxphone-0866-fe-1 kamailio_crash]# gdb kamailio core.kamailio.222447.pbxphone-0866-fe-1.company.name.1646903896 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-120.el7 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /root/kamailio_crash/kamailio...done. warning: exec file is newer than core file. [New LWP 222447] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamail'. Program terminated with signal 6, Aborted. #0 0x00007f04d6e16387 in raise () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install audit-libs-2.8.5-4.el7.x86_64 bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.176-5.el7.x86_64 glibc-2.17-325.el7_9.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-51.el7_9.x86_64 libacl-2.2.51-15.el7.x86_64 libattr-2.4.46-13.el7.x86_64 libcap-2.22-11.el7.x86_64 libcap-ng-0.7.5-4.el7.x86_64 libcom_err-1.42.9-19.el7.x86_64 libdb-5.3.21-25.el7.x86_64 libffi-3.0.13-19.el7.x86_64 libgcc-4.8.5-44.el7.x86_64 libselinux-2.5-15.el7.x86_64 libstdc++-4.8.5-44.el7.x86_64 libunistring-0.9.3-9.el7.x86_64 libuuid-2.23.2-65.el7_9.1.x86_64 libxml2-2.9.1-6.el7_9.6.x86_64 lm_sensors-libs-3.4.0-8.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 net-snmp-agent-libs-5.7.2-49.el7_9.1.x86_64 net-snmp-libs-5.7.2-49.el7_9.1.x86_64 nspr-4.32.0-1.el7_9.x86_64 nss-3.67.0-4.el7_9.x86_64 nss-softokn-freebl-3.67.0-3.el7_9.x86_64 nss-util-3.67.0-1.el7_9.x86_64 openssl-libs-1.0.2k-24.el7_9.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-299.el7_9.x86_64 popt-1.13-16.el7.x86_64 python-libs-2.7.5-90.el7.x86_64 rpm-libs-4.11.3-48.el7_9.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-19.el7_9.x86_64 (gdb) bt full #0 0x00007f04d6e16387 in raise () from /lib64/libc.so.6 No symbol table info available. #1 0x00007f04d6e17a78 in abort () from /lib64/libc.so.6 No symbol table info available. #2 0x000000000041e330 in sig_alarm_abort (signo=14) at main.c:693 __FUNCTION__ = "sig_alarm_abort" #3 <signal handler called> No symbol table info available. #4 0x00007f04d6ed8e27 in syscall () from /lib64/libc.so.6 No symbol table info available. #5 0x00000000004fa848 in futex_get (lock=0x7f042fe11ca0) at core/mem/../futexlock.h:121 v = 1 i = 0 #6 0x0000000000520dda in tcpconn_send_put (c=0x7f042fe11c98, buf=0x7ffd8be90c90 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"..., len=302, send_flags=...) at core/tcp_main.c:2422 tmp = 0x0 fd = 0 response = {140726950757744, 0} n = 32516 do_close_fd = 1 rest_buf = 0x12fe11e10 <Address 0x12fe11e10 out of bounds> t_buf = 0x0 rest_len = 32516 t_len = 794570504 resp = 139658225205256 t_send_flags = {f = 1296, blst_imask = 35817} fd_cache_e = 0x0 use_fd_cache = 1 __FUNCTION__ = "tcpconn_send_put" #7 0x00000000005203e7 in tcp_send (dst=0x7ffd8be90a20, from=0x0, buf=0x7ffd8be90c90 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"..., len=302) at core/tcp_main.c:2323 c = 0x7f042fe11c98 ip = {af = 2, len = 4, u = {addrl = {1765950034, 4294967295}, addr32 = {1765950034, 0, 4294967295, 0}, addr16 = {16978, 26946, 0, 0, 65535, 65535, 0, 0}, addr = "RBBi\000\000\000\000\377\377\377\377\000\000\000"}} port = 46050 fd = 4 response = {139655957499112, 0} n = 0 con_lifetime = 9600 try_local_port = 5061 rest_buf = 0x69d0c8 <dns_a_get_he+86> "\353CH\213Eؾ\001" t_buf = 0x7ffd8be906e0 "\020\a\351\213\375\177" rest_len = 0 t_len = 0 resp = 139655957577728 t_send_flags = {f = 19200, blst_imask = 180} __FUNCTION__ = "tcp_send" #8 0x00007f04b2aa0ceb in ul_ka_send (kamsg=0x7ffd8be90c80, kadst=0x7ffd8be90a20) at ul_keepalive.c:271 __FUNCTION__ = "ul_ka_send" #9 0x00007f04b2aa0b58 in ul_ka_urecord (ur=0x7f042fca3340) at ul_keepalive.c:239 uc = 0x7f042fca3458 kabuf = "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name >;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"... ---Type <return> to continue, or q <return> to quit--- kabuf_len = 302 kamsg = { s = 0x7ffd8be90c90 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.name:5061;branch=z9hG4bKx.14.1.0\r\nFrom: <sip:pbxphone@tone.company.name>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21c-c"..., len = 302} vaddr = {s = 0x7f04b6e74008 "tone.company.name", len = 12} vport = {s = 0x7f04b6e740f8 "5061", len = 4} sdst = {s = 0x7f042fca3770 "sip:82.66.66.105:46050;transport=tls", len = 36} sproto = {s = 0x81bb4b "TLS", len = 3} duri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x7f042fca3774 "82.66.66.105:46050;transport=tls", len = 12}, port = {s = 0x7f042fca3781 "46050;transport=tls", len = 5}, params = {s = 0x7f042fca3787 "transport=tls", len = 13}, sip_params = {s = 0x7f042fca3787 "transport=tls", len = 13}, headers = {s = 0x0, len = 0}, port_no = 46050, proto = 3, type = SIP_URI_T, flags = (unknown: 0), transport = {s = 0x7f042fca3787 "transport=tls", len = 13}, 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 = 0x7f042fca3791 "tls", len = 3}, 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}} dproto = 3 '\003' he = 0xb88fa0 <he.5288> ssock = 0x7f04b6e73d98 idst = {send_sock = 0x7f04b6e73d98, to = {s = {sa_family = 2, sa_data = "\263\342RBBi\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 58035, sin_addr = {s_addr = 1765950034}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 58035, sin6_flowinfo = 1765950034, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = { 0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 2, __ss_padding = "\263\342RBBi", '\000' <repeats 111 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 3 '\003', proto_pad0 = 0 '\000', proto_pad1 = 0} bcnt = 1 aortype = 0 i = 5 tv = {tv_sec = 1646903836, tv_usec = 846405} tnow = 1646903836 __FUNCTION__ = "ul_ka_urecord" #10 0x00007f04b2a6bb99 in mem_timer_udomain (_d=0x7f042f9c5388, istart=0, istep=1) at udomain.c:1234 ptr = 0x7f042fca3340 t = 0x1ea00000000 i = 28 #11 0x00007f04b2a9dd0e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:966 res = 0 ptr = 0x7f042f9c5288 #12 0x00007f04b2a7ac5a in destroy () at usrloc_mod.c:498 __FUNCTION__ = "destroy" #13 0x00000000005934ee in destroy_modules () at core/sr_module.c:772 t = 0x7f04b6e83828 foo = 0x7f04b6e83338 __FUNCTION__ = "destroy_modules" #14 0x000000000041cda7 in cleanup (show_status=1) at main.c:569 memlog = 0 __FUNCTION__ = "cleanup" #15 0x000000000041e682 in shutdown_children (sig=15, show_status=1) at main.c:712 __FUNCTION__ = "shutdown_children" #16 0x000000000042172a in handle_sigs () at main.c:812 chld = 0 chld_status = 139 any_chld_stopped = 1 memlog = 798824048 __FUNCTION__ = "handle_sigs" #17 0x000000000042babb in main_loop () at main.c:1845 i = 4 pid = 222489 si = 0x0 si_desc = "udp receiver child=3 sock=128.141.18.117:5060 ( tone.company.name:5060)\000\000\000\000\000\000\000\000\200S/\004\177\000\000\000\000\000\000\000\000\000\000\220Q\234/\004\177\000\000\200\034\351\2---Type <return> to continue, or q <return> to quit--- 13\375\177\000\000\000R\234/\004\177\000\000B\227Z\257\004\177\000\000\300\023\006\267\004\177\000" nrprocs = 4 woneinit = 1 __FUNCTION__ = "main_loop" #18 0x000000000043423e in main (argc=9, argv=0x7ffd8be921f8) at main.c:2942 cfg_stream = 0xf6a040 c = -1 r = 0 tmp = 0x7ffd8be935cc "" tmp_len = 1472 port = 960 proto = 32516 ahost = 0x0 aport = 0 options = 0x7d7ef8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1413124693 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffd8be920b0 p = 0xf0b5ff <Address 0xf0b5ff out of bounds> st = {st_dev = 20, st_ino = 16553497, st_nlink = 2, st_mode = 16832, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1646819986, tv_nsec = 228520360}, st_mtim = {tv_sec = 1646903531, tv_nsec = 836398702}, st_ctim = {tv_sec = 1646903531, tv_nsec = 836398702}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 392 times>... option_index = 0 long_options = {{name = 0x7da22f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d56d4 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7da234 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7da23a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7da240 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7da249 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7da253 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7da25d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7da268 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7da271 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7da27c "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x7da282 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" 2nd CORE backtrace [root@pbxphone-0866-fe-1 kamailio_crash]# gdb kamailio core.kamailio.222460.pbxphone-0866-fe-1.company.domain.1646903834 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-120.el7 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /root/kamailio_crash/kamailio...done. warning: exec file is newer than core file. [New LWP 222460] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 0x00007f04af578716 in tls_complete_init (c=0x7f042fe11c98) at tls_server.c:236 236 data->ssl = SSL_new(dom->ctx[process_no]); Missing separate debuginfos, use: debuginfo-install audit-libs-2.8.5-4.el7.x86_64 bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.176-5.el7.x86_64 glibc-2.17-325.el7_9.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-51.el7_9.x86_64 libacl-2.2.51-15.el7.x86_64 libattr-2.4.46-13.el7.x86_64 libcap-2.22-11.el7.x86_64 libcap-ng-0.7.5-4.el7.x86_64 libcom_err-1.42.9-19.el7.x86_64 libdb-5.3.21-25.el7.x86_64 libffi-3.0.13-19.el7.x86_64 libgcc-4.8.5-44.el7.x86_64 libselinux-2.5-15.el7.x86_64 libstdc++-4.8.5-44.el7.x86_64 libunistring-0.9.3-9.el7.x86_64 libuuid-2.23.2-65.el7_9.1.x86_64 libxml2-2.9.1-6.el7_9.6.x86_64 lm_sensors-libs-3.4.0-8.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 net-snmp-agent-libs-5.7.2-49.el7_9.1.x86_64 net-snmp-libs-5.7.2-49.el7_9.1.x86_64 nspr-4.32.0-1.el7_9.x86_64 nss-3.67.0-4.el7_9.x86_64 nss-softokn-freebl-3.67.0-3.el7_9.x86_64 nss-util-3.67.0-1.el7_9.x86_64 openssl-libs-1.0.2k-24.el7_9.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-299.el7_9.x86_64 popt-1.13-16.el7.x86_64 python-libs-2.7.5-90.el7.x86_64 rpm-libs-4.11.3-48.el7_9.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-19.el7_9.x86_64 (gdb) bt full #0 0x00007f04af578716 in tls_complete_init (c=0x7f042fe11c98) at tls_server.c:236 dom = 0x7f042fbcccf8 data = 0x7f042f96dec0 cfg = 0x7f042f96dec0 state = S_TLS_CONNECTING sname = 0x0 srvid = 0x0 __FUNCTION__ = "tls_complete_init" #1 0x00007f04af578d07 in tls_fix_connection_unsafe (c=0x7f042fe11c98) at tls_server.c:297 __FUNCTION__ = "tls_fix_connection_unsafe" #2 0x00007f04af580784 in tls_h_encode_f (c=0x7f042fe11c98, pbuf=0x7ffd8be909e0, plen=0x7ffd8be909d8, rest_buf=0x7ffd8be909e8, rest_len=0x7ffd8be909dc, send_flags=0x7ffd8be909c0) at tls_server.c:772 n = 0 offs = 0 ssl = 0x1 tls_c = 0x7f042fe11ca0 wr_buf = "\026\003\001\002\000\001\000\001\374\003\003\276,'P^\252j\027\021S\351\252\021\342(~\347\216\351\332Hh!\vb\024\361\324+\003\177\a\000\000\254\300\060\300,\300(\300$\300\024\300\n\000\245\000\243\000\241\000\237\000k\000j\000i\000h\000\071\000\070\000\067\000\066\000\210\000\207\000\206\000\205\300\062\300.\300*\300&\300\017\300\005\000\235\000=\000\065\000\204\300/\300+\300'\300#\300\023\300\t\000\244\000\242\000\240\000\236\000g\000@ \000?\000>\000\063\000\062\000\061\000\060\000\232\000\231\000\230\000\227\000E\000D\000C\000B\300\061\300-\300)\300%\300\016\300\004\000\234\000<\000/\000\226\000A\300\022\300\b\000\026\000\023\000\020\000\r\300\r\300\003"... rd = {buf = 0x18 <Address 0x18 out of bounds>, pos = 22930096, used = 1771061403, size = -1947662592} wr = {buf = 0x4b6e74a40 <Address 0x4b6e74a40 out of bounds>, pos = -1226359400, used = 32516, size = -1562640382} ssl_error = 0 err_src = 0x7f04af5afc09 "TLS write:" ip_buf = "\000\000\000\000\000\000\000\000\230\034\341/\004\177\000\000\000\000\000\000\000\000\000\000\370@ \347\266\001\000\000\000\340\b\351\213\375\177\000\000f\211Q\000\000\000\000\000\000\000\000\000\001\000\000\000\060\000\000\000\060\000\000" buf = 0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain >;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"... len = 308 x = 4 __FUNCTION__ = "tls_h_encode_f" #3 0x000000000051e729 in tcp_send (dst=0x7ffd8be90d60, from=0x0, buf=0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"..., len=308) at core/tcp_main.c:2093 c = 0x7f042fe11c98 ip = {af = 2, len = 4, u = {addrl = {1765950034, 4294967295}, addr32 = {1765950034, 0, 4294967295, 0}, addr16 = {16978, 26946, 0, 0, 65535, 65535, 0, 0}, addr = "RBBi\000\000\000\000\377\377\377\377\000\000\000"}} port = 46050 fd = 22 response = {139655966192064, 0} n = 0 con_lifetime = 9600 try_local_port = 5061 rest_buf = 0x0 t_buf = 0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain >;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"... rest_len = 0 t_len = 308 resp = 140726950756352 t_send_flags = {f = 0, blst_imask = 0} __FUNCTION__ = "tcp_send" #4 0x00007f04b2aa0ceb in ul_ka_send (kamsg=0x7ffd8be90fc0, kadst=0x7ffd8be90d60) at ul_keepalive.c:271 __FUNCTION__ = "ul_ka_send" #5 0x00007f04b2aa0b58 in ul_ka_urecord (ur=0x7f042fca3340) at ul_keepalive.c:239 uc = 0x7f042fca3458 kabuf = "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-2c-31dd20---Type <return> to continue, or q <return> to quit--- 1c-6229c21a"... kabuf_len = 308 kamsg = { s = 0x7ffd8be90fd0 "OPTIONS sip:73054@82.66.66.105:46050;transport=tls SIP/2.0\r\nVia: SIP/2.0/TLS tone.company.domain:5061;branch=z9hG4bKx.2854.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-2c-31dd201c-6229c21a"..., len = 308} vaddr = {s = 0x7f04b6e74008 "tone.company.domain", len = 12} vport = {s = 0x7f04b6e740f8 "5061", len = 4} sdst = {s = 0x7f042fca3770 "sip:82.66.66.105:46050;transport=tls", len = 36} sproto = {s = 0x81bb4b "TLS", len = 3} duri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x7f042fca3774 "82.66.66.105:46050;transport=tls", len = 12}, port = {s = 0x7f042fca3781 "46050;transport=tls", len = 5}, params = {s = 0x7f042fca3787 "transport=tls", len = 13}, sip_params = {s = 0x7f042fca3787 "transport=tls", len = 13}, headers = {s = 0x0, len = 0}, port_no = 46050, proto = 3, type = SIP_URI_T, flags = (unknown: 0), transport = {s = 0x7f042fca3787 "transport=tls", len = 13}, 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 = 0x7f042fca3791 "tls", len = 3}, 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}} dproto = 3 '\003' he = 0xb88fa0 <he.5288> ssock = 0x7f04b6e73d98 idst = {send_sock = 0x7f04b6e73d98, to = {s = {sa_family = 2, sa_data = "\263\342RBBi\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 58035, sin_addr = {s_addr = 1765950034}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 58035, sin6_flowinfo = 1765950034, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = { 0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 2, __ss_padding = "\263\342RBBi", '\000' <repeats 111 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 3 '\003', proto_pad0 = 0 '\000', proto_pad1 = 0} bcnt = 1 aortype = 0 i = 5 tv = {tv_sec = 1646903834, tv_usec = 680904} tnow = 1646903834 __FUNCTION__ = "ul_ka_urecord" #6 0x00007f04b2a6bb99 in mem_timer_udomain (_d=0x7f042f9c5388, istart=0, istep=1) at udomain.c:1234 ptr = 0x7f042fca3340 t = 0x7f0429d60470 <str_record_route_col> i = 28 #7 0x00007f04b2a9dd0e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:966 res = 0 ptr = 0x7f042f9c5288 #8 0x00007f04b2a7b019 in ul_core_timer (ticks=1791845, param=0x0) at usrloc_mod.c:527 __FUNCTION__ = "ul_core_timer" #9 0x00000000004f1e69 in compat_old_handler (ti=28669522, tl=0x7f042f82e060, data=0x7f042f82e060) at core/timer.c:980 t = 0x7f042f82e060 #10 0x00000000004f29a5 in slow_timer_main () at core/timer.c:1105 n = 12 ret = 4294967295 tl = 0x7f042f82e060 i = 308 __FUNCTION__ = "slow_timer_main" #11 0x000000000042a707 in main_loop () at main.c:1765 i = 4 pid = 0 si = 0x0 si_desc = "udp receiver child=3 sock=128.141.18.117:5060 (tone.company.domain:5060)\000\000\000\000\000\000\000\000\200S/\004\177\000\000\000\000\000\000\000\000\000\000\220Q\234/\004\177\000\000\200\034\351\213\375\177\000\000\000R\234/\004\177\000\000B\227Z\257\004\177\000\000\300\023\006\267\004\177\000" nrprocs = 4 woneinit = 1 __FUNCTION__ = "main_loop" #12 0x000000000043423e in main (argc=9, argv=0x7ffd8be921f8) at main.c:2942 cfg_stream = 0xf6a040 ---Type <return> to continue, or q <return> to quit--- c = -1 r = 0 tmp = 0x7ffd8be935cc "" tmp_len = 1472 port = 960 proto = 32516 ahost = 0x0 aport = 0 options = 0x7d7ef8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1413124693 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffd8be920b0 p = 0xf0b5ff <Address 0xf0b5ff out of bounds> st = {st_dev = 20, st_ino = 16553497, st_nlink = 2, st_mode = 16832, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1646819986, tv_nsec = 228520360}, st_mtim = {tv_sec = 1646903531, tv_nsec = 836398702}, st_ctim = {tv_sec = 1646903531, tv_nsec = 836398702}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 392 times>... option_index = 0 long_options = {{name = 0x7da22f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d56d4 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7da234 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7da23a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7da240 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7da249 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7da253 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7da25d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7da268 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7da271 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7da27c "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x7da282 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" 3rd CORE: [root@pbxphone-0866-fe-1 kamailio_crash]# gdb kamailio core.kamailio.276013.pbxphone-0866-fe-1.company.domain.1646903518 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-120.el7 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /root/kamailio_crash/kamailio...done. warning: exec file is newer than core file. [New LWP 276013] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 0x00007f7cfc309478 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at ../../core/atomic/atomic_x86.h:224 224 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) Missing separate debuginfos, use: debuginfo-install audit-libs-2.8.5-4.el7.x86_64 bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.176-5.el7.x86_64 glibc-2.17-325.el7_9.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-51.el7_9.x86_64 libacl-2.2.51-15.el7.x86_64 libattr-2.4.46-13.el7.x86_64 libcap-2.22-11.el7.x86_64 libcap-ng-0.7.5-4.el7.x86_64 libcom_err-1.42.9-19.el7.x86_64 libdb-5.3.21-25.el7.x86_64 libffi-3.0.13-19.el7.x86_64 libgcc-4.8.5-44.el7.x86_64 libselinux-2.5-15.el7.x86_64 libstdc++-4.8.5-44.el7.x86_64 libunistring-0.9.3-9.el7.x86_64 libuuid-2.23.2-65.el7_9.1.x86_64 libxml2-2.9.1-6.el7_9.6.x86_64 lm_sensors-libs-3.4.0-8.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 net-snmp-agent-libs-5.7.2-49.el7_9.1.x86_64 net-snmp-libs-5.7.2-49.el7_9.1.x86_64 nspr-4.32.0-1.el7_9.x86_64 nss-3.67.0-4.el7_9.x86_64 nss-softokn-freebl-3.67.0-3.el7_9.x86_64 nss-util-3.67.0-1.el7_9.x86_64 openssl-libs-1.0.2k-24.el7_9.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-299.el7_9.x86_64 popt-1.13-16.el7.x86_64 python-libs-2.7.5-90.el7.x86_64 rpm-libs-4.11.3-48.el7_9.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-19.el7_9.x86_64 (gdb) bt full #0 0x00007f7cfc309478 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at ../../core/atomic/atomic_x86.h:224 ret = 0 #1 0x00007f7cfc3094c4 in futex_get (lock=0x0) at ../../core/futexlock.h:99 v = 32636 i = 1024 #2 0x00007f7cfc310461 in wsconn_get (id=992) at ws_conn.c:452 id_hash = 992 wsc = 0x100000001 __FUNCTION__ = "wsconn_get" #3 0x00007f7cfc324d71 in ws_frame_transmit (evp=0x7ffcf5f31830) at ws_frame.c:728 wsev = 0x7ffcf5f31870 frame = {fin = 1, rsv1 = 0, rsv2 = 0, rsv3 = 0, opcode = 1, mask = 0, payload_len = 301, masking_key = "\000\000\000", payload_data = 0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., wsc = 0x0} __FUNCTION__ = "ws_frame_transmit" #4 0x00000000004f3762 in sr_event_exec (type=11, evp=0x7ffcf5f31830) at core/events.c:296 ret = 0 i = 50304 #5 0x0000000000544edf in wss_send (dst=0x7ffcf5f31c30, buf=0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., len=301) at core/tcp_main.c:5209 port = 50304 ip = {af = 2, len = 4, u = {addrl = {3232828610, 140724434836000}, addr32 = {3232828610, 0, 4126349856, 32764}, addr16 = {3266, 49329, 0, 0, 6688, 62963, 32764, 0}, addr = "\302\f\261\300\000\000\000\000 \032\363\365\374\177\000"}} from = 0x0 local_addr = {s = {sa_family = 19200, sa_data = "\264\000\000\000\000\000\360\034\363\365\374\177\000"}, sin = {sin_family = 19200, sin_port = 180, sin_addr = {s_addr = 0}, sin_zero = "\360\034\363\365\374\177\000"}, sin6 = {sin6_family = 19200, sin6_port = 180, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "\360\034\363\365\374\177\000\000\000\000\272\001}\177\000", __u6_addr16 = {7408, 62963, 32764, 0, 0, 442, 32637, 0}, __u6_addr32 = {4126350576, 32764, 28966912, 32637}}}, sin6_scope_id = 0}, sas = {ss_family = 19200, __ss_padding = "\264\000\000\000\000\000\360\034\363\365\374\177\000\000\000\000\272\001}\177\000\000\000\000\000\000\000\000\000\000\360\030\363\365\374\177\000\000\310\320i\000\000\000\000\000\300}\271\001}\177\000\000\000\000\000\000\000\000\000\000\340\271A\000\004\000\000\000\360\362\200\000\000\000\000\000\377\377\377\377", '\000' <repeats 12 times>, " \031\363\365\374\177\000\000\200\200\025\251}\177\000\000\000\000\000\000\000\000\000", __ss_align = 140724434836720}} con = 0x7f7d03e4c500 wsev = {type = 11, buf = 0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., len = 301, id = 992} evp = {data = 0x7ffcf5f31870, obuf = {s = 0x0, len = 0}, rcv = 0x0, dst = 0x0, req = 0x0, rpl = 0x0, rplcode = 0, mode = 0} ret = 0 __FUNCTION__ = "wss_send" #6 0x00007f7d84a54c47 in ul_ka_send (kamsg=0x7ffcf5f31e90, kadst=0x7ffcf5f31c30) at ul_keepalive.c:259 __FUNCTION__ = "ul_ka_send" #7 0x00007f7d84a54b58 in ul_ka_urecord (ur=0x7f7d04fcb8f8) at ul_keepalive.c:239 uc = 0x7f7d04fcba10 kabuf = "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain >;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"... kabuf_len = 301 kamsg = { s = 0x7ffcf5f31ea0 "OPTIONS sip:73904@7nprmvqiutl9.invalid;transport=wss SIP/2.0\r\nVia: SIP/2.0/WS tone.company.domain:8089;branch=z9hG4bKx.1.1.0\r\nFrom: <sip:pbxphone@tone.company.domain>;tag=uloc-6228658a-43655-44-31dd2800-6229c0de-6"..., len = 301} vaddr = {s = 0x7f7d88e28a40 "tone.company.domain", len = 12} vport = {s = 0x7f7d88e28b30 "8089", len = 4} sdst = {s = 0x7f7d051d1318 "sip:194.12.177.192:50304;transport=ws", len = 37} sproto = {s = 0x81bb5d "WS", len = 2} duri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x7f7d051d131c "194.12.177.192:50304;transport=ws", len = 14}, port = {s = 0x7f7d051d132b "50304;transport=ws", len = 5}, params = {s = 0x7f7d051d1331 "transport=ws", len = 12}, sip_params = {s = 0x7f7d051d1331 "transport=ws", len = 12}, headers = {s = 0x0, len = 0}, port_no = 50304, proto = 5, type = SIP_URI_T, ---Type <return> to continue, or q <return> to quit--- flags = (unknown: 0), transport = {s = 0x7f7d051d1331 "transport=ws", len = 12}, 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 = 0x7f7d051d133b "ws", len = 2}, 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}} dproto = 5 '\005' he = 0xb88fa0 <he.5288> ssock = 0x7f7d88e287d0 idst = {send_sock = 0x7f7d88e287d0, to = {s = {sa_family = 2, sa_data = "Ā\302\f\261\300\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 32964, sin_addr = {s_addr = 3232828610}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 32964, sin6_flowinfo = 3232828610, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = { 0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 2, __ss_padding = "Ā\302\f\261\300", '\000' <repeats 111 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 5 '\005', proto_pad0 = 0 '\000', proto_pad1 = 0} bcnt = 1 aortype = 0 i = 5 tv = {tv_sec = 1646903518, tv_usec = 435121} tnow = 1646903518 __FUNCTION__ = "ul_ka_urecord" #8 0x00007f7d84a1fb99 in mem_timer_udomain (_d=0x7f7d01990460, istart=0, istep=1) at udomain.c:1234 ptr = 0x7f7d04fcb8f8 t = 0x1ea00000000 i = 0 #9 0x00007f7d84a51d0e in synchronize_all_udomains (istart=0, istep=1) at dlist.c:966 res = 0 ptr = 0x7f7d01990360 #10 0x00007f7d84a2ec5a in destroy () at usrloc_mod.c:498 __FUNCTION__ = "destroy" #11 0x00000000005934ee in destroy_modules () at core/sr_module.c:772 t = 0x7f7d88e37828 foo = 0x7f7d88e37338 __FUNCTION__ = "destroy_modules" #12 0x000000000041cda7 in cleanup (show_status=1) at main.c:569 memlog = 0 __FUNCTION__ = "cleanup" #13 0x000000000041e682 in shutdown_children (sig=15, show_status=1) at main.c:712 __FUNCTION__ = "shutdown_children" #14 0x000000000041f1f7 in handle_sigs () at main.c:743 chld = 0 chld_status = 0 any_chld_stopped = 0 memlog = 26855240 __FUNCTION__ = "handle_sigs" #15 0x000000000042babb in main_loop () at main.c:1845 i = 4 pid = 276058 si = 0x0 si_desc = "udp receiver child=3 sock=128.141.18.117:5060 (tone.company.domain:5060)\000\000\000\000\000\000\000\000\300N\001}\177\000\000\000\000\000\000\000\000\000\000h\002\231\001}\177\000\000\220.\363\365\374\177\000\000\330\002\231\001}\177\000\000B\327U\201}\177\000\000\300S\001\211}\177\000" nrprocs = 4 woneinit = 1 __FUNCTION__ = "main_loop" #16 0x000000000043423e in main (argc=9, argv=0x7ffcf5f33408) at main.c:2942 cfg_stream = 0xf06040 c = -1 r = 0 tmp = 0x7ffcf5f34eff "" tmp_len = 1472 ---Type <return> to continue, or q <return> to quit--- port = 960 proto = 32637 ahost = 0x0 aport = 0 options = 0x7d7ef8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 626051482 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffcf5f332c0 p = 0xf0b5ff "" st = {st_dev = 20, st_ino = 16553497, st_nlink = 2, st_mode = 16832, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1614937626, tv_nsec = 968330983}, st_mtim = {tv_sec = 1646814601, tv_nsec = 999994878}, st_ctim = {tv_sec = 1646814601, tv_nsec = 999994878}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 392 times>... option_index = 0 long_options = {{name = 0x7da22f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d56d4 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7da234 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7da23a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7da240 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7da249 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7da253 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7da25d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7da268 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7da271 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7da27c "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x7da282 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" Is it a right place to report or better to make an issue on github? Thanks in advance! -- Best regards, Igor

1 0

Kamailio v5.4.8 Released
by Daniel-Constantin Mierla 09 Mar '22

09 Mar '22

Hello, Kamailio SIP Server v5.4.8 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.8. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.4 being release out of it. For more details about version 5.4.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/03/kamailio-v5-4-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online March 28-31, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

Releasing Kamailio v.5.4.8
by Daniel-Constantin Mierla 09 Mar '22

09 Mar '22

Hello, I am planning to release a new version from branch 5.4, respectively 5.4.8, sometime next week, likely on Wednesday or Thursday (Mar 9/10, 2022). If anyone is aware of issues not yet reported to bug tracker or missing backports, report them in order to try to get the fixes in this release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online March 28-31, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

2 3

rtpengine - SRTP <> RTP missing a=crypto
by Rhys Hanrahan 09 Mar '22

09 Mar '22

Hi Everyone, I have Kamailio sitting between MS Teams and Asterisk, and using rtpengine to terminate SRTP on Kamailio so that all my internal traffic is unencrypted. My current config works fine for inbound calls where I initiate the INVITE and Teams responds, but if Teams sends the INVITE I am having an issue where SRTP cannot finish negotiating. Non SRTP calls work fine with RTPEngine as well, so it's just the RTP to SRTP I am struggling with. According to this I believe I must pass a=crypto in response to the INVITE which also has a=crypto: https://www.dialogic.com/-/media/1f8b54b43087407d9c2b38846c5c2cb5.ashx?h=40… You can see that in the initial invite from Teams, I get RTP/SAVP with a=crypto, but I do not send one in my OK response after 183 Session In Progress. As below - I am wondering if it's because not all audio channels seem to be getting swapped to SAVP? I'd like to do a generic SRTP <> RTP bridge config (I've tried below). However, I am not 100% sure on how to detect when to swap between AVP and SAVP, so I've also tried just doing rtpengine_manage() and relying on other code to swap between SAVP or AVP *only* when going to/from Teams to keep it simple. I also tried both with and without "replace-origin replace-session-connection ICE=remove" but I still get the same behaviour in all cases. Any advice appreciated, as this is my first time dealing with SRTP (and rtpengine). Feeling very stuck. Thanks! branch_route[MANAGE_BRANCH] { ... route(NATMANAGE); route(HANDLE_SRTP); } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } route(HANDLE_SRTP); } route[HANDLE_SRTP] { if (!has_body("application/sdp")) { return; } rtpengine_manage(); return; # As a test, just do rtpengine_manage() and set SAVP/AVP elsewhere. Same behaviour. # Handle bridging of RTP and SRTP # Inbound traffic to SBC should be converted from SRTP to RTP if (proto==TLS) { rtpengine_manage("RTP/AVP"); # Outbound traffic destined to a TLS destination should be converted from RTP to SRTP } else if ($ru =~ "transport=tls") { rtpengine_manage("RTP/SAVP"); } } # INVITE from teams rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP"); # INVITE to teams rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/SAVP"); INVITE sip:+614xxxx@rh.sbc-syd-01.teams.xxxx:5061;user=phone;transport=tls SIP/2.0^M ... v=0^M o=- 57931 0 IN IP4 127.0.0.1^M s=session^M c=IN IP4 52.113.76.53^M b=CT:10000000^M t=0 0^M m=audio 51398 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118^M c=IN IP4 52.113.76.53^M a=rtcp:51399^M a=ice-ufrag:C8ss^M a=ice-pwd:2bV9D6GcXF5f8m0px/wufQD/^M a=rtcp-mux^M a=candidate:1 1 UDP 2130706431 52.113.76.53 51398 typ srflx raddr 10.0.32.179 rport 51398^M a=candidate:1 2 UDP 2130705918 52.113.76.53 51399 typ srflx raddr 10.0.32.179 rport 51399^M a=candidate:2 1 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=candidate:2 2 tcp-act 2121006078 52.113.76.53 49152 typ srflx raddr 10.0.32.179 rport 49152^M a=label:main-audio^M a=mid:1^M a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:geUHLB1mshmnI5hN83bnO57Hbdm2i7dD14sDAnpA|2^31^M a=sendrecv^M a=rtpmap:104 SILK/16000^M a=rtpmap:9 G722/8000^M a=rtpmap:103 SILK/8000^M a=rtpmap:111 SIREN/16000^M a=fmtp:111 bitrate=16000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:97 RED/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=rtpmap:13 CN/8000^M a=rtpmap:118 CN/16000^M a=ptime:20^M I correctly convert to/from RTP/AVP and RTP/SAVP for the 183 Session in progress. It is RTP/SAVP before going to Teams: SIP/2.0 183 Session Progress^M ... v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M m=audio 37820 RTP/SAVP 9 8 0 101^M a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M But then when I send the OK after the 183, I am setting RTP/SAVP before sending to MS Teams, but not setting a=crypto: Also note that I can see there are _some_ channels still as RTP/AVP so maybe this is part of the issue. SIP/2.0 200 OK^M ... v=0^M o=- 57931 2 IN IP4 1.2.3.4^M s=NexusOne^M c=IN IP4 1.2.3.4^M t=0 0^M m=audio 37820 RTP/SAVP 9 8 0 101^M a=maxptime:150^M a=mid:1^M a=rtpmap:9 G722/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:37821^M a=ptime:20^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M m=audio 0 RTP/AVP 104 9 103 111 18 0 8 97 101 13 118^M Rhys Hanrahan | Chief Information Officer e: rhys(a)nexusone.com.au<mailto:rhys@nexusone.com.au> [www.nexusone.com.au]<http://www.nexusone.com.au/> [signature_1116663581] <http://www.fusiontech.com.au/> NEXUS ONE | FUSION TECHNOLOGY SOLUTIONS p: 1800 NEXUS1 (1800 639 871) or 1800 565 845 | a: Suite 12.03 Level 12, 227 Elizabeth Street, Sydney NSW 2000 www.nexusone.com.au<http://www.nexusone.com.au/> | www.fusiontech.com.au<http://www.fusiontech.com.au/> The information in this email and any accompanying attachments may contain; a. Confidential information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; b. Legally privileged information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; and or c. Copyright material Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties. If you have received this email in error, please notify the sender immediately and delete this message. Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd does not accept any responsibility for loss or damage arising from the use or distribution of this email. Please consider the environment before printing this email.

4 13

Kamailio + KEMI + topos + change_reply_status
by Marrold 09 Mar '22

09 Mar '22

Hi, I'm using Kamailio 5.5.3 with a KEMI / Python based configuration on Debian 11. In my reply route I am trying to convert a 183 to a 180 with: KSR.textopsx.change_reply_status(180, "Ringing") I've noticed with TOPOS enabled, it doesn't work and I still see a 183 go out. But with it disabled, it works as expected and it's converted to a 180. I get the same behaviour with / without msg_apply_changes() Config looks like this: def ksr_onreply_manage(self, msg): KSR.info(f"ksr_onreply_manage\r\n") scode = KSR.pv.get("$rs") if KSR.permissions.allow_source_address_group()<0: if scode == 183: KSR.info(f"ksr_onreply_manage: Converting 183 to 180\r\n") KSR.textopsx.change_reply_status(180, "Ringing") KSR.textopsx.msg_apply_changes() if scode>100 and scode<299 : self.ksr_route_natmanage(msg) return 1 The debug log doesn't reveal anything too obvious, I can share if its of any use. Does anyone have any suggestions? Thanks Matthew

2 2

kamailio.cfg : behind the NAT (listen and advertise) understanding
by Youssef Boujraf 07 Mar '22

07 Mar '22

Hi Dear I am trying to understand how "listen" and "advertise" work for sip signaling purpose ( corebooks : listen <http://www.kamailio.org/wiki/cookbooks/5.4.x/core#listen> and advertise ) Flow : kamailio behind NAT using softphones on private and public networks. - kamailio : Network : IPv4_private 192.168.1.6 / IPv4_public kamailio.cfg : listen=udp: IPv4_private: 5060 advertise IPv4_public : 5060 - uac (softphone) : zoiper, microsip, cisco spa508g USE CASE : I 'd like to force all softphone on the same local network to reach kamailio through a public ip address. ??? QUESTION : Do I need to REGISTER on IPv4_private address of Kamailio OR REGISTER on IPv4_public address of kamailio or DEFINE another "listen" with another PORT WITHOUT "advertise" for softphone on same kamailio network ? listen=udp: IPv4_private: 5060 advertise IPv4_public : 5060 listen=udp: IPv4_private: 5066 Do I need to use "kamctl ps" to verify which interface is listening on port ? Thanks in advance Best Regards, Youssef BOUJRAF Email: yboujraf(a)gmail.com

3 5

simple GUI for Kamailio as a SBC - could be paid
by Krzysztof Drewicz 05 Mar '22

05 Mar '22

Hello, We searched for a simple GUI that could drive a kamailio - a bunch of trunks, some custom logic, tried to use open source Sipwise C5 - a way too big and heavy. Any hints ? links ? Mabe a paid subscription? Our goal is - to have a GUI that could - put a version number to configuration, and have a simple way to audit changes - only to log - what was done, when and from which management account. vi + svn/git - it's working but this is no GUI, ACLs in sqlite, also work, but - need to have a ssh console, keep files in some repository - well this is simply inconvenient when you have more than 5 trunks with registration etc... BR, -- *Krzysztof Drewicz* Senior Infrastructure Administrator CLUDO | ul. Grochowska 306/308, 03-840 Warszawa t+48221223977 kdrewicz(a)cludo.pl | www.cludo.pl <http://www.cludo.pl/>

5 7

OS-assigned TLS source Port#?
by Matthias Urlichs 02 Mar '22

02 Mar '22

Hi, My Kamailio installation originates TLS connections using source port 5061. Is there a way to teach it not to do that? Seting tcp_reuse_port=no doesn't have any effect. -- -- Matthias Urlichs

2 2

ndb_redis - redis-cluster error
by Maharaja Azhagiah 02 Mar '22

02 Mar '22

Hi All, I am new to Kamailio and testing location server using kamailio and redis-cluster. My setup is simple: All SIP REGISTER messages will be forwarded to this location-server. The server checks for AUTH and then needs to save the usrloc in redis-cluster. I have hiredis and hiredis-cluster libraries installed I am using ndb_redis to support redis-cluster with below configuration: modparam("ndb_redis", "server", "name=172.17.0.3:7000 ;addr=172.17.0.3;port=7000") modparam("ndb_redis", "server", "name=172.17.0.4:7001 ;addr=172.17.0.4;port=7001") modparam("ndb_redis", "server", "name=172.17.0.5:7002 ;addr=172.17.0.5;port=7002") modparam("ndb_redis", "server", "name=172.17.0.6:7003 ;addr=172.17.0.6;port=7003") modparam("ndb_redis", "server", "name=172.17.0.7:7004 ;addr=172.17.0.7;port=7004") modparam("ndb_redis", "server", "name=172.17.0.8:7005 ;addr=172.17.0.8;port=7005") modparam("ndb_redis", "cluster", 1) If SIP method is REGISTER, Kamailio sends redis_cmd to store as below: redis_cmd("172.17.0.3:7000", "sadd %s %s", "location:usrdom::437780082: mazhagiah.sip.zang.io", "location:entry::$xavp(ulrcd=>ruid)", "r"); However, redis-cluster replies with MOVED message, but kamailio / ndb_redis throughs error as below: 0(3915) DEBUG: ndb_redis [redis_client.c:391]: redisc_get_server(): Hash 2590462363 (172.17.0.3:7000) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2591109923 (172.17.0.8:7005) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590232843 (172.17.0.7:7004) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590175467 (172.17.0.6:7003) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590347603 (172.17.0.5:7002) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590290227 (172.17.0.4:7001) 0(3915) DEBUG: ndb_redis [redis_client.c:395]: redisc_get_server(): Entry 2590462363 (172.17.0.3:7000) 0(3915) DEBUG: ndb_redis [redis_client.c:896]: redisc_exec(): rsrv->ctxRedis = 0x558e4d739210 0(3915) DEBUG: ndb_redis [redis_client.c:903]: redisc_exec(): rsrv->ctxRedis = 0x558e4d739210 0(3915) DEBUG: ndb_redis [redis_client.c:957]: redisc_exec(): rpl->rplRedis->type:6 0(3915) ERROR: ndb_redis [redis_client.c:959]: redisc_exec(): Redis error:MOVED 10082 172.17.0.4:7001 0(3915) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-621e8438-f4b-b] (11 / 19) 0(3915) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-621e8438-f4b-b 0(3915) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-621e8438-f4b-c] (12 / 19) Please help if I am missing anything with my configuration or library. Thanks Raja

2 1

Problem with auth_ephemeral and parse_uri(): bad port in uri
by Vlasis Chatzistavrou 01 Mar '22

01 Mar '22

Hello, I have a problem with Kamailio 5.4.6 and auth_ephemeral. I have the following in the Kamailio configuration loadmodule "auth_ephemeral" modparam( "auth_ephemeral", "sha_algorithm", 3 ) modparam( "auth_ephemeral", "username_format", 0 ) modparam( "auth_ephemeral", "secret", 1234 ) as per https://kamailio.org/docs/modules/4.1.x/modules/auth_ephemeral.html#auth_ep… and registrations fail. In the logs we see: Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:777]: check_parse_uris(): looking up From header Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:817]: check_parse_uris(): parsing From URI Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: DEBUG: {1 545 REGISTER rhaqgafd7boteg24jp5db0} <core> [core/parser/parse_uri.c:1296]: parse_uri(): bad port in uri (error at char 5 in state 2) parsed: <sip:3518929:16411>(17) /<sip:3518929:1641150726@192.168.2.99> (35) Jan 2 18:21:10 enswitch43 /sbin/kamailio[37501]: WARNING: {1 545 REGISTER rhaqgafd7boteg24jp5db0} sanity [sanity.c:820]: check_parse_uris(): failed to parse From uri Apparently Kamailio is confused by the timestamp following the username separated by the : character. The REGISTER message is below: REGISTER sip:192.168.2.99 SIP/2.0 Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK5452321 Max-Forwards: 70 To: "3518929" <sip:3518929:1641148397@192.168.2.99> From: "3518929" <sip:3518929:1641148397@192.168.2.99>;tag=ht76o8b2b6 Call-ID: phkj9mi2n3s3ju7uu3qq2f CSeq: 274 REGISTER Contact: <sip:edh7mmti@192.0.2.202;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:ca5e9372-dfa1-459a-b6ba-4398d23bd896>";expires=300 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, outbound User-Agent: Raspberry Phone (SipJS - 0.11.6) Content-Length: 0 and Kamailio parses it as sip:<username>:<port> instead of sip:<username>:<timestamp>. Is this a bug that should be reported or is there any setting that I am missing?

4 9

t_on_failure => what variables being reset to which values?
by Benoît Panizzon 01 Mar '22

01 Mar '22

Hi List We t_on_failure re-route calls to another destination in case one fails for example announcement servers... Before sending to the announcement server I do add some extra header to define which announcement in which language should be played: append_hf("IMP-annc: $avp(announcecode)\r\n"); append_hf("IMP-mandate: $avp(mandate)\r\n"); append_hf("IMP-language: $avp(callerlang)\r\n"); Also $rU is being set prior to sending the call to the announcement server. Now one of our announcement servers became unavailable and t_on_failure was triggered. The 2nd announcement servers was receiving the calls without any of the additional header and with the 'inbound' $ru without the altered $rU https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_on_failu… I'm a bit confused by that, as this states the URI is reset to the value it had on 'relaying'. Is this on 'inbound replaying' or on 'outbound relaying' when route(RELAY) was called to send the call TO the first annoucement server? Is there a bit more specific information on that behaviour? -- 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

KEMI jsdt CommonJS-based module
by Ian Carlson 01 Mar '22

01 Mar '22

Greetings, I'm new to Kamailio and working to add Kamailio to an existing Asterisk solution. I'm attracted to the idea of routing via KEMI jsdt since our Asterisk ARI is using nodejs. Playing with KEMI jsdt I was trying to use require() to break up the monolithic routing script. It seems that Kamailio uses Ducktape 2 without module-node or module-ducktape, requiring all routing in a single file? I'd appreciate any suggested practices for managing Kamailio configurations. Cheers, Ian

2 4

RTP Engine early media issue one to many
by Filippo Graziola 28 Feb '22

28 Feb '22

Hi everyone, in my environment (kamailio - http API for retrieve destinations - rtpengine) it is used a lot parallel forking, the flow is as follows: A -> INVITE -> Kamailio -> http API/rtjson / t_relay -> Kamailio -> client B / C / D (only one accepting the call) Kamailio loops to itself to have a parallel forking, it involves rtpengine correctly. Both audio and video work during call. The issue comes with one of destinations sending 183 with SDP, the caller will send then a video preview as early media before call is established. I noticed that only the first one "asking" for early media is receiving this video preview (sometimes none of clients requesting it), all others are ignored by caller. This is a case of 1-to-many, I searched a lot about this but it seems at least not working or not supported, could you confirm that there is no configuration trick possible to make work this early media to many destinations? Another issue in above call flow, is that if client B is receiving correctly this early media video and client C answers the call (200OK) client C does not see any video, audio working and video is completely missing from rtp, only a reinvite makes possible to see video. Is this a client side only issue (so only solution is the reinvite) or there is something missing on my kamailio configuration? Thanks Filippo

1 0

ACK and DNS failover
by Julien Klingenmeyer 25 Feb '22

25 Feb '22

Hello, I use DNS failover feature for routing some calls, and I wonder about its limitations. I noticed that INVITE and ReINVITE requests are correctly routed based on SRV priority/weight in case of failure, but ACK requests do not use them. Let’s say I have Kamailio A relaying requests to a pool of Kamailios B1 and B2 in TLS. DNS records are the below ones. kamailiob.net. 60 NAPTR 20 100 S SIPS+D2T _sips._tcp.kamailiob.net. _sips._tcp.kamailiob.net. 60 SRV 1 10 5061 kamailiob-1.net. _sips._tcp.kamailiob.net. 60 SRV 2 10 5061 kamailiob-2.net. If B1 is down, I expect requests being relayed to B2 (because of priority 2 in the SRV record). * For initial INVITEs: R-URI is sip:kamailiob.net * If B1 is down, the request is retried to B2 => OK * For in-dialog requests: Route header in incoming request is sip:kamailiob.net so it is set as next destination with loose_route function * If B1 is down, the request is retried to B2 for ReINVITE => OK * But once the ACK comes in, with the same Route header (sip:kamailiob.net), Kamailio tries to send it to B1 only (no retries to B2, and even no retransmissions to B1 either). * When B1 is back UP again a few seconds later, Kamailio does not try to relay the ACK again. The ACK request is attempted to be retransmitted only once to B1. It fails and no more retries after that. Is it the expected behavior? Or is it something misconfigured on Kamailio A? Or is it a TLS connection issue? Here is the DNS configuration on Kamailio A (TM module is enabled): dns_try_naptr=yes use_dns_failover=yes use_dns_cache=yes dns_srv_lb=yes enable_tls=yes tcp_max_connections=4096 tls_max_connections=4096 tcp_reuse_port=yes tcp_connect_timeout=10 tcp_send_timeout=10 tcp_connection_lifetime=3600 I had a look into the DNS tutorial (https://github.com/kamailio/kamailio/blob/master/doc/tutorials/dns.txt) without finding any hint about this. If anyone has already played with SIP DNS failover in Kamailio, your help would be appreciated, thanks! Julien

3 4

kamailio and certificate renewal
by Aymeric Moizard 25 Feb '22

25 Feb '22

Hi all, I'm switching my kamailio to use Let's Encrypt certificates. Do I need to force a restart of kamailio when the certificate is renewed or will this be automatic? The doc indicates that reload is possible (but not advised) but it looks to be for a "configuration change" (tls.cfg file change) and not for certificate change. Maybe certificates are automatically reloaded? Regards Aymeric -- Antisip - http://www.antisip.com

2 1

Re-using an existing TLS connection
by Matthias Urlichs 25 Feb '22

25 Feb '22

Hello, My problem: our provider opens a TLS connection to us. They want us to use this channel for outgoing calls, instead of opening a new SIP connection to their server. Is there a way to teach Kamailio to do that? -- -- Matthias Urlichs

3 3

Info: SIP resources collection
by Daniel-Constantin Mierla 24 Feb '22

24 Feb '22

Hello, over the 20 years I used various SIP applications, libraries and tools but after a while I forgot many of them, therefore I tried to collect many in a list hosted on github at: * https://github.com/miconda/sip-resources Hoping it proves useful for some of you! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

4 6

kamailio : how to xlog the value of "myself" ?
by Youssef Boujraf 24 Feb '22

24 Feb '22

Dear I 'd like to xlog the value of the "myself" ? if(uri==myself) { log("the request is for local processing\n"); };Best Regards, -- Youssef BOUJRAF

3 3

dispatcher.reload
by David Villasmil 24 Feb '22

24 Feb '22

Hello guys, We're dispatching with hash over user, if i execute dispatcher.reload <http://www.kamailio.net/docs/modules/5.0.x/modules/dispatcher.html#dispatch…> does the dispatcher list gets calculated? i.e. user A is always dispatched to box3, if i reload, is the hash recalculated and will probably be dispatched differently? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 4

Dispatcher Load Balance with SIP Trunk
by warawich 24 Feb '22

24 Feb '22

Hello, I would like to do a load balance between Asterisk SIP trunks. You can see a diagram from this link: https://drive.google.com/file/d/1Qy66L5rQCfxysYQpSd2-ek_8-by0T8PR/view?usp=… SIP Packets capture log: https://drive.google.com/file/d/1CHGUOwoRDAC93MMBtyfa8gBISKIVC-ng/view?usp=… Details 1. Asterisk1 makes SIP trunk connection with Kamailio. 2. Kamailio makes SIP trunk connection with Asterisk2 and Asterisk3 3. Caller register SIP phone with Asterisk1 Caller extension = 8002 4. Asterisk2 makes a blind transfer to Kamailio. (Call to 8009 then transfer to ARI APP) exten = 8009,1,Transfer(SIP/3802@<kamailio public IP>) 5. ARI app extension is 3802 My problem is that Kamailio LB only works when I try to connect with Asterisk1 and Asterisk2 or Asterisk1 and Asterisk3. If I have two Asterisks in dispatcher.list, it doesn't work and it appears *SIP/2.0 401 Unauthorized* in sip packets capture log. Kamailio version: 5.5.2 Kamailio.cfg modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 3) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } dispatcher.list # setid(int) destination(sip uri) flags(int,opt) priority(int,opt) attrs(str,opt) 1 sip:10.148.0.31:5123 1 sip:10.148.0.44:5123 Thanks, Warawich

2 2

SQLOPS sql_query_async() not functioning in an async manner?
by Ben Kaufman 23 Feb '22

23 Feb '22

I'm not sure if this is by design or not. I have things I want to log to a SQL DB, but it's not nearly as important as call handling, so the sql_query_async() function looked ideal, but in testing, if the DB is not responsive, the call processing gets blocked. In the example below, I'm using just a short hostname which isn't resolvable, but if I have an IP address there that isn't really assigned (for example if the DB server is down), the same blocking behavior is exhibited, but the error is slightly different (can't connect instead of unknown host). Here's my config. This specific simple config was tested on Kamailio 5.5, but I observe similar behavior (in a more complicated config) in the dev branch as well. - - - - #!KAMAILIO async_workers=2 loadmodule "db_mysql" loadmodule "sqlops" loadmodule "sl" loadmodule "xlog" loadmodule "pv" force_rport=yes ## Host sql does not resolve. modparam("sqlops","sqlcon","cb=>mysql://kamailio:kamailiorw@sql/kamailio") ## But start anyways. modparam("sqlops", "connect_mode", 1) request_route { sl_send_reply("100", "Trying"); ## Prevent re-transmissions from the UAC xlog("L_NOTICE", "Time before sql_query_async(): [$TV(Sn)]\n"); sql_query_async("cb", "INSERT INTO table1 (col) VALUES ('val')"); xlog("L_NOTICE", "Time before sl_send_reply(): [$TV(Sn)]\n"); sl_send_reply("404", "Not found"); } Here's the log if I send a simple INVITE with SIPp. Note the timestamps 9(16) NOTICE: <script>: Time before sql_query_async(): [1645574694.546165] 9(16) ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'sql' (-3) 9(16) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 9(16) ERROR: sqlops [sql_api.c:190]: sql_reconnect(): failed to connect to the database [cb] 9(16) ERROR: sqlops [sqlops.c:232]: sql_check_connection(): failed to connect to database 9(16) ERROR: sqlops [sqlops.c:264]: sql_query_async(): invalid connection to database 9(16) NOTICE: <script>: Time before sl_send_reply(): [1645574699.552440] Ben Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

4 4

kamailio :?==?utf-8?q? ROUTE[AUTH] - IPAUTH - use case
by Youssef Boujraf 23 Feb '22

23 Feb '22

Dear, I enabled two softphone with ADDRESS. - codec1 : ip 192.168.30.241/32 5060 - codec2 : ip 192.168.30.242/32 5060 I enable one softphone with SUBSCRIBER - codec3 : is registered (login/passwd) codec4 is not part of the domain, is not subriber and address. Kamailio proxy - kamailio : ip 192.168.30.240/32 5060 In the kamailio.cfg ROUTE[AUTH]# IP authorization and user authenticationroute[AUTH] {#!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return;} Regarding the condition #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; }#!endif USE CASE : If codec4 (not registered or same domain) tries to INVITE codec3 (in subscriber) the codec, an INVITE MESSAGE is sent and the codec3 could ACK. - But I don't want to allow the INVITE, I 'd like to not allow the codec4 to reach codec3. - But if codec1 or codec2 tries to reach codec3, kamailio will allow the INVITE. Best Regards, -- Youssef BOUJRAF

1 0

Issue with Authenticating outbound Call
by Victor ku 22 Feb '22

22 Feb '22

Hi, I'm new to sip and kamailio, I have recently setup a docker container based on kamailio/kamailio:5.5.0-stretch to check things out. While I was trying to authenticate an outbound call to a 3rd party sip provider from my kamailio container, I noticed there are times that the 2nd invite with the authorization header that's supposed to be sent from the failure route doesn't get sent out. There's no error recorded in the log nor does wireshark picking up any reply to sip phone or message to 3rd party sip provider. Anyone know what might have caused this? Some other things to note: Kamailio does sent out the INVITE with authorization header and authenticated successfully when my sip phone sent the sip request with FROM ip set to the 3rd party sip server's ip. When I change the sip phone to send INVITE with FROM Domain using kamailio container's ip, I'm able to see header is being built from the trace but no INVITE with authorization header is being sent. Changing the FROM uri using $fu or uac_replace_from() does help either. ---------------------------------------------------------------------------------------------------------------- > > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} uac [auth_hdr.c:408]: > build_authorization_hdr(): hdr is <Authorization: Digest > username="xxxxxxxx", realm="xxx.xxx.xxx.xxx", nonce="07f4a5779568", > uri="sip:+0123456789@xxx.xxx.xxx.xxx", qop=auth, nc=00000001, > cnonce="3782993246", response="d5ebd0ef00d44db2292c7b8508fa23bc", > algorithm=MD5 > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1026 a=16 > n=if > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1021 a=24 > n=t_relay > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} <core> [core/msg_translator.c:1797]: > check_boundaries(): no multi-part body > ... > ... > kamailio-kamailio-1 | 14(22) exec: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} *** cfgtrace:dbg_cfg_trace(): > failure_route=[MANAGE_FAILURE] c=[/etc/kamailio/kamailio.cfg] l=1030 a=2 > n=exit > kamailio-kamailio-1 | 14(22) DEBUG: {1 11126 INVITE > 6cda7ecf9aed4154bfb409224b84a876} dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483650 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1564]: > t_should_relay_response(): rps discarded - uas status: 100 > kamailio-kamailio-1 | 14(22) DEBUG: tm [t_reply.c:1924]: relay_reply(): > reply status=2 branch=0, save=0, relay=-1 icode=0 msg status=401 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:596]: > receive_msg(): reply-route executed in: 1753 usec > kamailio-kamailio-1 | 14(22) DEBUG: dialog [dlg_var.c:97]: > cb_dlg_locals_reset(): resetting the local dialog shortcuts on script > callback: 2147483652 > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/usr_avp.c:637]: > destroy_avp_list(): destroying list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/xavp.c:541]: > xavp_destroy_list(): destroying xavp list (nil) > kamailio-kamailio-1 | 14(22) DEBUG: <core> [core/receive.c:606]: > receive_msg(): cleaning up ---------------------------------------------- package capture ---------------------------- 192.168.0.165 xxx.xxx.xxx.xxx SIP/SDP 1383 Request: INVITE > sip:+0123456789@ xxx.xxx.xxx.xxx | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 585 Status: 100 Trying | > xxx.xxx.xxx.xxx 192.168.0.165 SIP 820 Status: 401 Unauthorized | > 192.168.0.165 xxx.xxx.xxx.xxx SIP 396 Request: ACK sip:+0123456789 > @ xxx.xxx.xxx.xxx | ------------------------------------------nothing more till sip phone cancel ----------------- # ----------------- below is failure route config --------------------- > failure_route[MANAGE_FAILURE] { > xlog("~~~~~ MANAGE_FAILURE ($rm) ($rs-$rr) (from $fu) (to $tu) (req :$ru) > (dest $du)\r\n"); if (t_is_canceled()) { exit; } xlog("Checking status code\r\n"); #!ifdef WITH_UAC if(t_check_status("401|407")) { if (isflagset(7)) { t_reply("503","Authentication failed"); exit; } $avp(auser) = "xxxxxxxxxx"; $avp(apass) = "xxxxxxxxxx; if(uac_auth()) { setflag(7); xlog("after uac_auth $rm (from $fu)(to $tu)(req $ru)(dest $du)\r\n"); if(!t_relay()) { xlog("Failed to relay ($rm) ($rs-$rr) (from $fu)(to $tu)(req $ru)\r\n"); t_reply("500","Error"); } } else { xlog("uac_auth() not available\r\n"); t_reply("503","Authentication invalid"); } exit; } #!endif } > # ----------- end failure route config --------------------

2 2

reinvite with 2 video lines in SDP
by Filippo Graziola 22 Feb '22

22 Feb '22

Hi all, I'm facing a strange behaviour from a client (iOS application based on pjsip library) which is doing as follow when receiving a call (on phone locked): - accept the call rejecting video (m=video 0...) in SDP - reinvite to negotiate again video media Specifically the reinvite passing through kamailio (+rtpengine) has this SDP body after rtpengine offer (the client use TLS so I can't see the original message): v=0 o=- 3854443139 3854443142 IN IP4 PUBLIC_RTP_ENGINE_IP s=pjmedia b=AS:386 t=0 0 a=X-nat:0 m=audio 46734 RTP/AVP 8 101 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:96000 a=ssrc:1456418416 cname:441b90ac6db72f22 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:46735 m=video 0 RTP/AVP 96 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 m=video 46776 RTP/AVP 98 c=IN IP4 PUBLIC_RTP_ENGINE_IP b=TIAS:256000 a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=42e01e; packetization-mode=1 a=sendrecv a=rtcp:46777 from this I can understand that the client is doing something strange, because there is still the rejection line (I know that this question should go to the pjsip guys) but I would also like to know if it is possible to "clean" this sdp by kamailio. A few clients receiving this sdp are answering with only audio. Reading the docs, sdpops/textops modules allow modifying lines by prefix or media, but till now I have no luck to exclude only the line "m=video 0 RTP/AVP 96" and his attributes without touching the correct video request. Could someone please give me some hints on it?

2 2

Enable switch redirect to other SIP server source
by bayu p 21 Feb '22

21 Feb '22

Dear Kamailio Expert, Could you help me switching redirect sip server to another sip server ? As described below architecure. I have some uac_redirect module to implement , but kamailio cannot perform running states. Pleas help to reditect sip server to another sip server. Thanks & Best regards,

2 1

kamailio success story
by Valentin Christoph 21 Feb '22

21 Feb '22

Hi all, We had to connect to an SBC, which was not compliant to RFC 3261. It was urgent, because management was waiting for first call. Within minutes, I could implement a workaround in kamailio routing configuration: ## CV experimental 2022-02-17: fake Supported HF if (is_present_hf("Require") && !is_present_hf("Supported")) { append_hf("Supported: $hdr(Require)\r\n", "Require"); msg_apply_changes(); } Thank you, kamailio! All the best Christoph

2 1

SIP server port service cannot be changed from default value (5060)
by bayu p 18 Feb '22

18 Feb '22

Hello Experts, Please help me to change port service from default 5060 port into other port due to security reason. Since I have change port in kamailio.cfg file, but services cannot fly normally that shown client unable login online (status not online). [image: image] <https://user-images.githubusercontent.com/76675592/154027407-24ab423b-302b-…> Besides, I need to run the kamailio services after install default kamailio, I have not found where procedure found in this installation process on github.com/kamailio. Could you please share if there another steps of compiling Kamailio source, install, deploy process, and running command for this specific installation process? Thanks Gb

4 9

Kamailio v5.5.4 Released
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, Kamailio SIP Server v5.5.4 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.4. For more details about version 5.5.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/02/kamailio-v5-5-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Releasing Kamailio v5.5.4
by Daniel-Constantin Mierla 17 Feb '22

17 Feb '22

Hello, I am considering to release Kamailio v5.5.4 later this week, likely on Thursday, Feb 17, 2022. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

Info: sipexer v1.0.0 - sip cli tool
by Daniel-Constantin Mierla 16 Feb '22

16 Feb '22

Hello, I want to announce the availability of sipexer v1.0.0 - a sip cli tool that can facilitate testing and monitoring of SIP signalling systems. It tries to have a modern approach, with a flexible templating system, supporting both IPv4 and IPv6 with all the common transport layers, respectively UDP, TCP, TLS and WebSocket (for WebRTC). The project can be found at: * https://github.com/miconda/sipexer It is written in Go language for better portability, binaries for Linux, MacOS and Windows are made available for download in the release page: * https://github.com/miconda/sipexer/releases/tag/v1.0.0 Among its features: * send OPTIONS request (quick SIP ping to check if server is alive) * do registration and un-registration with customized expires value and contact URI * authentication with plain or HA1 passwords * set custom SIP headers * template system for building SIP requests * fields in the templates can be set via command line parameters or a JSON file * variables for setting field values (e.g., random number, data, time, environment variables, uuid, random string, …) * simulate SIP calls at signalling layer (INVITE-wait-BYE) * respond to requests coming during SIP calls (e.g., OPTIONS keepalives) * send instant messages with SIP MESSAGE requests * color output mode for easier troubleshooting * support for many transport layers: IPv4 and IPv6, UDP, TCP, TLS and WebSocket (for WebRTC) * send SIP requests of any type (e.g., INFO, SUBSCRIBE, NOTIFY, …) One usage example that could ease the testing of Kamailio is initiating registrations or simulating calls over WebSocket without the need of having a JavaScript soft phone application running in a web browser. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

10 12

allow_source_address_group in failure route
by Marrold 16 Feb '22

16 Feb '22

Hello, I'm using Kamailio as a proxy between several voice gateways and if the call fails for whatever reason (far end is busy etc) I am forwarding to an Asterisk box to play an announcement. Within the failure route I am using allow_source_address_group() to ascertain if the failure response was from a Voice Gateway or Asterisk itself, however it appears to return the group for the first branch when the reply is from the second branch to Asterisk. To handle the failure I am adding some headers, and setting a new destination indirectly with ds_select_dst before t_relay. I am using 5.5.3 on Debian 11, using KEMI / Python. Does anyone have any ideas? Thanks Matthew

2 1

Missing contact in 180 ringing causes wrong routing
by Xuo Guoto 15 Feb '22

15 Feb '22

Hello all, I have a kamailio ( kamailio 5.5.3 (x86_64/linux)) with asterisk (Asterisk 16.2.1~dfsg-2ubuntu1) with following architecture ┌─────────────┐ ┌─────────────┐ ┌─────────────────────┐ │ Asterisk ├───────┤ Kamailio │ 188.xxx.59.171 ───────── 103.xxx.xx.134 │ Linphone │ └─────────────┘ └─────────────┘ └─────────────────────┘ 10.10.1.13 10.10.1.9 In case the diagram is not displayed correctly its pasted at https://pastebin.com/b0UtE4Lj Registrations are handled by Kamailio and calls from linphone are routed to Asterisk. Similarly all calls originating in asterisk are routed over to Kamailio which in turn looks up AOR and routes the calls over to Linphone. Along with Linphone I also have a WebRTC endpoint at Kamailio. When calls are sent to WebRTC from asterisk, it gets correctly routed over to Asterisk, but if the client is Linphone the ACK of 200OK is not routed correctly to Linphone. I get this error in the logs: [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced tls:188.xxx.59.171:5061, to udp:188.xxx.59.171:5060) Digging into the difference between WebRTC and Linphone, I could see that in 1800 Ringing packet from WebRTC has a Contact headers while its missing in the packet from Linphone This is the 180 ringing from WebRTC: tag: rcv pid: 695615 process: 17 time: 1644860126.209155 date: Mon Feb 14 23:05:26 2022 proto: wss ipv4 srcip: 103.xxx.xx.134 srcport: 52508 dstip: 188.xxx.59.171 dstport: 443 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/WSS 188.xxx.59.171:443;branch=z9hG4bK730a.2cc7461b72f0c895e7cce5a0cedce4f7.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Content-Length: 0 While this is from Linphone tag: rcv pid: 695614 process: 16 time: 1644860097.970935 date: Mon Feb 14 23:04:57 2022 proto: tls ipv4 srcip: 103.xxx.xx.134 srcport: 25843 dstip: 188.xxx.59.171 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Via: SIP/2.0/TLS 188.xxx.59.171:5061;branch=z9hG4bKb5b1.d3231367383d07b1063098dcb35cc2b0.0 Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPjd16f1379-1be7-448e-894a-c42d96f63146 From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=cf7a7770-529d-4450-8bed-880382dce1dd To: <sip:user_jbjp@10.10.1.9>;tag=2ooYDia Call-ID: 8a025999-b096-4fe0-a623-9f6e6a69b8ae CSeq: 27838 INVITE User-Agent: Linphone Desktop/4.3.2 (Ubuntu 20.04.3 LTS, Qt 5.12.12) LinphoneCore/5.0.49 Supported: replaces, outbound, gruu Record-route: <sip:188.xxx.59.171:5061;transport=tls;r2=on;lr=on;did=3b9.b1b2> Record-route: <sip:10.10.1.9;r2=on;lr=on;did=3b9.b1b2> Content-Length: 0 When the 180 ringing is passed on to the Astrerisk it has the contact header modified with alias. tag: snd pid: 695615 process: 17 time: 1644860126.210189 date: Mon Feb 14 23:05:26 2022 proto: udp ipv4 srcip: 10.10.1.9 srcport: 5060 dstip: 10.10.1.13 dstport: 7000 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing Record-Route: <sip:188.xxx.59.171:443;transport=ws;r2=on;lr=on;did=ad6.8121> Record-Route: <sip:10.10.1.9;r2=on;lr=on;did=ad6.8121> Via: SIP/2.0/UDP 10.10.1.13:7000;received=10.10.1.13;rport=7000;branch=z9hG4bKPje4d8c7fc-3ecd-421c-b4bd-698097abb4c7 To: <sip:user_jbjd@10.10.1.9>;tag=611ql89jnj From: "Anonymous" <sip:anonymous@erx-staging-q01.mydomain.com>;tag=5589ec6e-8c68-44de-b03c-72751d9b9fa4 Call-ID: 895f3d45-5479-4603-9d38-6429ff77b06d CSeq: 16925 INVITE Contact: <sip:user_jbjd@erx-staging-q01.mydomain.com;alias=103.xxx.xx.134~52508~6;gr=urn:uuid:a8b474b8-1e18-43cb-ab17-f6bbe02c548d> Supported: gruu, outbound User-Agent: SIP.js/0.7.8 Content-Length: 0 Now, I am not sure if this is the reason for routing the packets to wrong destination, but this is all I could figure out. Any clue to get this resolved will be much appreciated. My Kamailio config is pasted at https://pastebin.com/qyRS0usw with regards, X. Sent with ProtonMail Secure Email.

1 0

get kamailio sip processes id
by Patrick Karton 14 Feb '22

14 Feb '22

hello, as we are all aware kamailio create a lot of processes for many tasks. im using kamailio in kubernetes and i would like to monitor only sip processes with top command. i only need a way to get only kamailio sip processes id (udp receiver that handle sip request) . but i dont knwo how to get it. top -c -p id1,id2,id3,id4 Thanks

3 5

SEMS license with kamailio and rtpengine
by Patrick Karton 14 Feb '22

14 Feb '22

Hello im using kamailio and rtpengine to build a custom SBC. i was using freeswitch as my b2bua but i noticed a lot of issues when cps and simultaneous increased so i tried sems and i have better results. But i do not really understand sems license so i want to know if i can use it. i read from sems github "It is licensed under dual license terms, the GPL (v2+) and proprietary license. This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed. For a license to use SEMS under non-GPL terms, please contact FRAFOS GmbH at info(a)frafos.com" so my question is if i want to use sems as b2bua with kamailio and rtpengine and make profit of it do i need to purchase a license each time i want to sell it to a customer ? Thanks.

7 39

Kamailio server behind NAT
by Tihomir Culjaga 13 Feb '22

13 Feb '22

Hi all I'm wondering what is best practice to configure kamailio for a setup shown below. Carriers only support interconnection over a private network so I need to terminate every separate network on kamailio machine (and on RTPEngine) . That will turn very quickly into an administration nightmare as we add more and more carriers. My plan is to terminate carrier private networks on a router and NAT the traffic to kamailio where it leads me to a kamailio behind NAT setup. Well, in theory set_advertised_address and set_advertised_port per message can do the trick... at least I hope. Also, some carriers require a SIP registration to activate SIP trunks. I do that using UAC module (remote registration) but here again I got a problem with contact address as you can set only one contact address (modparam("uac", "reg_contact_addr", "10.1.1.11:5060")). /--RTP 10000-19999-- RTPEngine1 -----RTP------+-- IPPBX1 carrier1 --- NAT ---| | | | | | carrier2 --- NAT ---+-----SIP 5060------ kamailio ----SIP---- Dispatcher | | | carrier3 --- NAT ---| | | \--RTP 20000-29999-- RTPEngine2 -----RTP------+-- IPPBX2 So bottom line, I'm wondering what would be the best practice for such a scenario ? That would spare me from hitting many pitfalls. Would really appreciate a hint. Regards, Tihomir.

3 3

Vo5G with open5gs
by Hornbuckle, Mark A 11 Feb '22

11 Feb '22

Hello, I have been trying to find documentation on Kamailio IMS and Vo5G support for open5gs. I wanted to ask you directly if Kamailio IMS does support Vo5G with open5gs? Thank you

1 0

resending same message from UE after receiving RP-ACK
by Eugene B 11 Feb '22

11 Feb '22

Hi, I trying to use smsops module as SMSC. When I sending message from UE I see re-sending same message after each RP-ACK(~10 attempt). What could be the problem? Thanks! pcap in attach. kamailio 5.3.2 -- *Yours faithfully,* *Eugene B.*

2 1

kamilio open5gs 5G SA compatible?
by Sandland, Seth P 11 Feb '22

11 Feb '22

Were looking to test a kamilio on Open5gs 5g stand alone ( non lte, non NSA) can you conform or provide documents or install instructions on how this is accomplished. I can not find much of any details if Kamilio support Open5Gs 5g SA and what features it does support on a 5g SA core. Seth Sandland Wireless Communications IT Engineer Pacific Northwest National Laboratory Tel: (509) 375-6842 Email: seth.sandland(a)pnnl.gov<mailto:seth.sandland@pnnl.gov> www.pnnl.gov<http://www.pnnl.gov/> [A close up of a logo Description automatically generated]

1 0

Publish XML and xpath
by Igor Olhovskiy 10 Feb '22

10 Feb '22

Hello, I'm trying to get state of endpoint from PUBLISH message in a most simple way using xmlops module Trying to make this $xml(body=>doc) = $rb; log("L_ERR", "$rU is now in state $xml(body=>xpath:/dialog-info/dialog/state)\n"); XML in PUBLISH itself <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:88881@dev.company"> <dialog id="v6qmEYf5yI" direction="initiator"> <state>confirmed</state> </dialog> </dialog-info> Actual text in log is ... <script>: 88881 is now in state Means it's empty. xmlops module is loaded with no parameters. Tried also string $xml(body=>xpath:/dialog-info/dialog/state/text()) -> same result What am I missing here? kamailio 5.4.6 -- Thanks in advance, Igor

2 5

KEMI KSR.async Python3
by Ahmed Moubayed 10 Feb '22

10 Feb '22

Hi, I'm trying to use the async module from a python3 Kemi script but it conflicts with the keyword async. Any suggestions on how it can be used? Is there an alternative to KSR.async.route()? Thanks, Ahmed

1 0

CDR preserve original call direction
by Ahmed Fouad 10 Feb '22

10 Feb '22

Hi, I've enabled ACC and CDR logging and it's working fine for the logging but I noticed something wrong in the CDR table , the call source and destinations are depending on the BYE message not the initial invite dialog ACC table +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | id | method | callid | time | src_ip | src_domain | src_user | dst_ouser | dst_domain | +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | 216 | BYE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:57:51 | x.x.x.x | domain.com | DST_CLID | SRC_CLID | domain.com | | 215 | INVITE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:55:17 | x.x.x.x | domain.com | SRC_CLID | DST_CLID | dst_domain.com | The INVITE is the correct one with SRCIP x.x.x.x and domain.com but the BYE reversed the order since the the DST_CLID who sent the BYE so it's now logged as the source and the final CDR log looks like +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | cdr_id | src_username | src_domain | dst_username | dst_domain | call_start_time | call_end_time +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | 195 | DST_CLID | domain.com | SRC_CLID | domain.com | 2022-02-09 06:55:12 | 2022-02-09 06:57:51 | the CDR as shown is reversed regarding the source/dst username I tried to enable the direction ACC direction flag but didn't help and here is my current paramters modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "cdrs") Can you please advise how to configure ACC/CDR to log the CDR with the correct initial INVITE call flow regardless of the BYE direction ?

3 5

REGISTER with http_async_client ?
by Cyril Ramière 10 Feb '22

10 Feb '22

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); } } ##### END SNIP Best regards!

4 15

Kamailio doc request
by Mustafa Rahman 10 Feb '22

10 Feb '22

2 1

Failure route with t_relay()
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I am trying to make a re-transmission mechanism from the failure route, I've tried to use $dp and $du as a relay, but variables are not accepted in t_relay(), When using ip and port straight forward t_relay("144.0.0.0", "1111"), I am getting: [t_fwd.c:1752]: t_forward_nonack(): no branches for forwarding [tm.c:1753]: _w_t_relay_to(): t_forward_noack failed Is there a way to re-transmit the call to the same carrier again? Thanks in advance, and have a great day, Max Kemel

2 1

Failure route mechanism with counter
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I would like to use a failure route with a retry mechanism, which provides a MAX_NUM_OF_RETRIES (max number of times reaching the failure route), Which can be achieved by a simple global counter, from what I've seen, the Counters module doesn't provide a GET function in script, What is the easiest solution for this kind of behavior? Thanks a lot, Max Kemel

2 1

Timers firing too soon
by Alberto Izquierdo 04 Feb '22

04 Feb '22

Hi all, I'm using Kamailio timers to execute a series of calls with a certain delay. I have found out that from time to time, the timers are triggered too soon, sometimes they are triggered 2/3 seconds before they should. I have read in the documentation that it is normal to have a bit of delay in some of the timers, but it doesn't mention anything about this behavior. Is this normal? Thank you.

1 0

Debian kamailio.init vs. kamailio.service
by Juha Heinanen 04 Feb '22

04 Feb '22

By comparing pkg/kamailio/deb/bullseye dir kamailio.init and kamailio.service files, it appears to me that they are not equal alternatives. Is this correct interpretation? -- Juha

1 0

Homer7 and Kamailio 5.3
by Angelo Sipper 03 Feb '22

03 Feb '22

Hi to all, We have a Kamailio 5.3 that sends all traffic to a homer7 using siptrace module with parameters force_send_sock and hep_mode_on,1 and hep_version,3 and trace_mode,1. Everything works fine getting all traffic to Homer7 but, in case Homer7 is not reachable, Kamailio then no longer accepts traffic or registrations and it is like it crashes. When we revive Homer7 and is it again reachable accepting traffic, Kamailio then works again fine without any restart or actions. Any ideas why this may be happening? Is there anything we need to add to the config to avoid this behavior as having Homer7 now is a single point of failure for Kamailio? Thank you! Kind Regards, Angelo

6 11

uac problem with 5.3.8
by Richard Robson 03 Feb '22

03 Feb '22

we have been running 5.3.4 with the uac_auth_mode() function ans it has been working we have upgraded to 5.3.8 and with the same script we are getting an error with the uac module saying it was unable to find the cmd auc_auth_mode I have tried this with uac_auth as well with the same result. it looks like the uac module is not loaded or is not loading but running a core.modules reports the uac module can anyone shed any light on this problem? is it a version pecific problem?

3 4

Updating $du for uac REGISTERs in event_route[tm:local-request]
by Raffaella Seuß 03 Feb '22

03 Feb '22

Hello all, I tried to update $du for REGISTERs sent by the uac module in event_route[tm:local-request]. (I want to overwrite the auth_proxy under some conditions.) However, the requests are still sent to the auth_proxy defined in uacreg. Am I missing something here, why doesn't it work? It's basically just: event_route[tm:local-request] { if (is_method("REGISTER")) { # ... $du = "sip:PROXY_IP:PROXY_PORT"; } } My Kamailio version is 5.5.2. With kind regards, Raffaella Seuß foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: raffaella.seuss(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

1 0

Siremis configuration
by Tim Mastarone 02 Feb '22

02 Feb '22

Hi, I am using Siremis v 5.3 with kamailio. My users can log into the Siremis site just fine but the field to enter their email address is not appearing. My sql queries seem to be working besides that. Does this feature need to be enabled? I would like to use email to confirm the user login with a confirmation code. (mysql Ver 15.1 Distrib 10.1.48-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2) Thank you, [cid:image001.png@01D81830.52C21EA0]<http://www.vikingelectronics.com/> TIM MASTARONE Associate Engineer 715-386-8861 x208 vikingelectronics.com<http://www.vikingelectronics.com/>

1 0

two SIP proxies
by Uwe Sigleur 31 Jan '22

31 Jan '22

2 1

tel2sip is giving unpredictable result
by Jyoti Bansal 31 Jan '22

31 Jan '22

hello, we are using kamailio 5.3 . while establishing a call the tel uri is getting converted into sip uri sometimes incorrectly . In the sip uri some extra characters are getting appended in the invite message. Can somebody please provide some guidance on this, attaching the pcap for reference(packet no. 8). Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

kamailio log dialog correlation
by marek 31 Jan '22

31 Jan '22

hi, is it possible create log of SIP dialog like https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging ? it looks like its possible like this xlog("L_INFO", "D$dlg(h_id) something to log \n"); but i must copy $dlg(h_id) in every xlog tried this modparam("xlog", "prefix", "D$dlg(h_id)") but it not works with variables any other tips/ideas? goal is to correlate logs for one dialog (call) thanks Marek

7 16

REGISTER with http_async_client?
by Cyril Ramière 31 Jan '22

31 Jan '22

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. It seems that my AUTH_REPLY route is called but considered as a "failure_route" 19(27) exec: *** cfgtrace:dbg_cfg_trace(): failure_route=[AUTH_REPLY] c=[/etc/kamailio/kamailio.cfg] l=665 a=16 n=if 17(23) NOTICE: <script>: [AUTH_REPLY] FAILURE ROUTE If I remove the http_async_client and try with route(AUTH_REPLY) then it is considered as request_route (but this is not a solution since I need to do this http request) Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); if(t_is_failure_route()) { xlog("L_NOTICE", "[AUTH_REPLY] FAILURE ROUTE \n\n"); } } } ##### END SNIP Best regards!

1 1

Use specific expires per User-Agent
by mayamatakeshi 31 Jan '22

31 Jan '22

Hi, I have a requirement to specify a REGISTER max_expires based on the User-Agent. I checked the docs and it seems there is no pseudo-variable or something like that would permit this. I would like to be able to do something like this: if($ua ~= "Some-UA") { $max_expires=100; } else { $max_expires=150; } save("location"); Is there some way to achieve this?

2 2

failed to find string from message body
by Juha Heinanen 29 Jan '22

29 Jan '22

I tried to check if message body contains a string using function call str_find("$rb", "string")) but it failed like this: Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1491]: pv_printf_mode(): no more space for spec value - printed:0 token:6880 buffer:4096 Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1503]: pv_printf_mode(): buffer overflow -- increase the buffer size... Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/sr_module.c:1698]: get_str_fparam(): Could not convert the PV-formated string to str Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: textops [textops.c:4457]: str_find_f(): cannot get first parameter Any suggestions on how to get it done properly? -- Juha

1 0

Question about siptrace destination/source port capture in HEPv3 mode
by Karsten Horsmann 29 Jan '22

29 Jan '22

Hello List, I observe an interesting behavior. We use siptrace as an active capture agent for QXIP HOMER/HEPIC installations. The destination port and the source port in the HEPv3 Header are not correct if you use transport tcp for example. My Question: is that intended or a bug and i should fill up an issue? On udp, it's filled with 5060 or whatever your listening directive says. I tested that with kamailio 5.5 and kamailio master. I attached an HEPv3 capture from two kamailios (master) speaking with each other. You need the https://github.com/sipcapture/hep-wireshark lua dissector to read the capture. 10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode) 192.168.50.4 (kamailio as registrar) Frame 1 shows an tcp transport register Request from 10.0.2.15 to 192.168.50.4 in HEPv3. The Source Port in the HEP3 Protocol is not correct (its an highport / 45419). Frame 4 shows the 401 answer and the Destination port is 0. Frame 7 also had this behavior. ----- params (nearly at the end): modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture agent id Limitation: 32-bit for HEPv3. modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need call siptrace flag or sip_trace() -- Mit freundlichen Grüßen / Kind Regards *Karsten Horsmann*

3 2

tel2sip is not working correctly
by Jyoti Bansal 29 Jan '22

29 Jan '22

Hello all, I am using kamailio code to establish calls with the real ue's. Our calls are getting established with an issue , whenever invite is send from A party to B party with teluri that tel uri is converted to sip uri. While converting to sip uri some extra bytes are getting added after user=phone and the call ends. Can someone help us on this issue of getting some extra bytes added in the sip uri? for example :- sip:405874112224411@ims.mnc123.mcc456.3gppnetwork.org;user=phone�� Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

1 0

Comunication between devices RTP/SRTP and viceversa
by Social Boh 28 Jan '22

28 Jan '22

I'm trying to configure Kamailio to permit communication between devices: some use TLS and SRTP some UDP and RTP with RTPEngine in the middle. A priori I don't know if the device support SRTP or RTP so in the routing I have to detect on the fly how to act. The problem is this: Call from RTP to SRTP device: 488 Not Acceptable here Call from SRTP to RTP device: 415 Unsupported Media Type Any hint? Thank you Regards -- --- I'm SoCIaL, MayBe

4 5

200 OK re transmission only on re-SUBSCRIBE
by Lewis Hutchinson 27 Jan '22

27 Jan '22

First of all I should start by saying this is my first post so go easy on me!! : ) and thank for everyone's efforts to share knowledge it has been invaluable for my kamailio learning. Build Details Version: kamailio 5.5.3 (x86_64/linux) Kamailio kemi python Scenario Relevant config def ksr_route_api_query(self, msg): KSR.log("info", "SUBSCRIBE API Query Starts") # Create new Transaction KSR.tm.t_newtran() async def main(): url = 'https://api.hidingtheurl.com/subscribe' payload = { "ruri": KSR.pv.get("$ru"), "from": KSR.pv.get("$fu"), "from_tag": KSR.pv.gete("$ft"), "to": KSR.pv.get("$tu"), "callid": KSR.pv.get("$ci"), "cseq": KSR.pv.get("$cs"), "contact": KSR.pv.gete("$ct"), "supported": KSR.pv.get("$hdr(Supported)"), "event": KSR.pv.get("$hdr(Event)"), "expires": KSR.pv.get("$hdr(Expires)"), "diversion": KSR.pv.get("$di"), "body": KSR.pv.get("$rb") } async with aiohttp.ClientSession() as session: async with session.post(url, json=payload) as response: if response.status == 0: KSR.xlog.xinfo(f"SUB Manager Timeout: {response.status}") KSR.tm.t_reply(408, "SUB Manager Timeout") elif response.status >= 500: KSR.xlog.xinfo(f"SUB Manager Down: {response.status}") KSR.tm.t_reply(500, "SUB Manager Down") elif response.status >= 400: KSR.xlog.xinfo(f"SUB Manager Error: {response.status}") KSR.tm.t_reply(400, "SUB Manager Error") else: if response.status == 200: KSR.xlog.xinfo(f"SUB Manager Success: {response.status}") KSR.tm.t_reply(200, "OK") KSR.log("info", "SUBSCRIBE API Query Ends") asyncio.run(main()) # API Query Ends Scenario Initial SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get receive 200OK response from API and send a 200OK on to Device SUBSCRIBE --------> <------------ 200 OK After Subscription Expiry re-SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get 200OK response from API and send a 200OK on to Device but also another 200OK a fraction of a second later... SUBSCRIBE --------> <------------ 200 OK <<<--------- 200 OK This happens for all following re-SUBSCRIBE's The 200 OK are identical, and it is a re transmission, but I can't work out why.. Relevant DEBUG is below Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: INFO: <script>: SUB Manager Success: 200 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_reply Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:287]: sr_apy_kemi_exec_func_ex(): params[2] for: t_reply are int-str: [200] [OK] Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (80.111.111.111, 80.111.111.111, 0) Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1763]: cleanup_uac_timers(): RETR/FR timers reset Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out - buf=0x7eff0cd50938: SIP/2.0 200 OK#015#012Via:... shmem=0x7eff09190ca8: SIP/2.0 200 OK#015#012Via: Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:648]: _reply_light(): finished Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_precheck_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_check_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7eff0cb18220: SIP/2.0 2..., shmem=0x7eff09190ca8: SIP/2.0 2 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:112]: sr_kemi_config_engine_python(): execution of route type 1 with no name returned 1 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/receive.c:514]: receive_msg(): request-route executed in: 38258 usec Everything works well as is but I want to clean up this issue or at least understand what is causing it. Lewis

2 6

Some problems with SIP TRACE
by Arnaud 27 Jan '22

27 Jan '22

Hi, I'm trying to use siptrace to getting copy of my SIP transactions, especially in TLS. Here's my configuration : *loadmodule "siptrace.so" modparam("siptrace", "trace_mode", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "duplicate_uri", "sip:MY_IP:5065") modparam("siptrace", "trace_local_ip", "MY_IP:5061") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_capture_id", 1) modparam("siptrace", "trace_flag",22); * And in my Routing Logic : *request_route {* *sip_trace();setflag(22);* *(...)* But I didn't receive anything on my listening port 5065 according my TCPDUMP. Status confirm me siptrace is working : *root@debian:/# kamcmd siptrace.status checkEnabled* I think I made a mistake in the configuration, but I don't know where. My second question is when I try to use sip_trace_mode("t"); I getting : failed to find command sip_trace_mode. Is it in an other module ? Many thanks for your help. Regards, Arnaud

2 1

Re: [SR-Users] sr-users Digest, Vol 200, Issue 25
by surendra tiwari 27 Jan '22

27 Jan '22

Hi Aditya, There are 2 options to achieve this: Option-1 use below param in auth_db module and while doing authentication change table name to your table user_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> domain_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> password_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> Option-2 create a new http service that will connect to kamailio via evapi and will be responsible for authentication and location service. On Wed, 26 Jan 2022 at 16:31, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Kamailio as a SIP registrar (Aditya Uppuluri) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Jan 2022 16:21:33 +0530 > From: Aditya Uppuluri <uppuluriaditya(a)gmail.com> > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Kamailio as a SIP registrar > Message-ID: > <CAKWpYb= > 9EABvpmbPhvCyt-roQaOkiJnZSh17vVu6Wk26PiBbeg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hello Team, > Thank you for the amazing product. I am a newbie to kamailio and currently > trying to use kamailio as a SIP Registrar (for a multidomain). Here is the > requirement > > 1. We already have a mysql database (name: VOIP) with a table named > VoipSubscribers which contains the userid, password and domain information > of all the users. > 2. We want the ability in kamailio to use the same VoipSubscribers table > for user registrations and also store the location information in the same > database (VOIP) > > May I know how to achieve this without altering the table schema? > Thank you in advance. > > Regards, > Aditya Uppuluri >

1 0

Kamailio as a SIP registrar
by Aditya Uppuluri 26 Jan '22

26 Jan '22

Hello Team, Thank you for the amazing product. I am a newbie to kamailio and currently trying to use kamailio as a SIP Registrar (for a multidomain). Here is the requirement 1. We already have a mysql database (name: VOIP) with a table named VoipSubscribers which contains the userid, password and domain information of all the users. 2. We want the ability in kamailio to use the same VoipSubscribers table for user registrations and also store the location information in the same database (VOIP) May I know how to achieve this without altering the table schema? Thank you in advance. Regards, Aditya Uppuluri

2 1

Append HF don't work in event_route
by Arnaud 24 Jan '22

24 Jan '22

Hello, I'm trying to connect kamailio to Microsoft Teams. Dispatch is correctly configured and sended. Following this tutorial : https://skalatan.de/en/blog/kamailio-sbc-teams, I added an Event_Route to add contact header to SIP OPTION from dispath to MS : event_route[tm:local-request] { xlog("*** tm:local-request *** "); if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { xlog("L_INFO", "Append new contact !\n"); append_hf("Contact: <sip:MY_SBC_NAME.COM:5061 ;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } When I check the log to see my sended frame, I don't see a contact header field : INFO: <script>: Append new contact ! INFO: <script>: Sent out tm request : OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS MY_SBC_NAME.COM:5061 ;branch=z9hG4bK4497.f3e5e754000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:MY_SBC_NAME.COM.fr>;tag=a76e006ace3802951ed860e08df964af-7971 CSeq: 10 OPTIONS Call-ID: 2e1278ea77796160-9837@MY_SBC_IP Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.2.1 (x86_64/linux)) According the documentation about textop, append_hf(txt[, hdr]) : This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE. So my question is append HF can be used in Event_route ? Is there any other way to add contact header ? Regards, Arnaud

1 0

g_hash_table_insert_internal: assertion 'hash_table != NULL' failed
by Muhammad Danish Moosa 23 Jan '22

23 Jan '22

Hi, I am running stress tests on kamalilio , at times i see this on screen (process:30959): GLib-CRITICAL **: 21:43:56.961: g_hash_table_insert_internal: assertion 'hash_table != NULL' failed version: kamailio 5.5.3 Kamailio does not crash and keeps accepting calls though. Is it anything alarming. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

1 0

dispatcher lb
by Gokan Atmaca 22 Jan '22

22 Jan '22

Hello I'm load balancing with Dispatcher. The call comes in but hangs up without picking up the phone. What could be the problem? The settings are as follows. dispatcher.list 1 sip:18.X.X.X.X:5060 0 0 maxload=20 1 sip:67.X.X.X.X:5060 0 0 maxload=20 # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 4.130000 STD: 0.398000 EST: 4.006000 MAX: 6 TIMEOUT: 0 } } DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 22.282000 STD: 2.345000 EST: 23.075000 MAX: 42 TIMEOUT: 0 } } } } } } -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

2 3

Effect of GRUU on the behavior of loose_route()
by Denys Pozniak 21 Jan '22

21 Jan '22

Hello! I ask for help in explaining the different behavior of loose_route() function in two similar cases (depending on GRUU parameter on the end device). The call path goes through a series of Proxies: Proxy Registrar (use Path) -> Proxy Edge -> PhonerLite (GRUU on/off). If the GRUU parameter is enabled on the PhonerLite, then the in-dialog SIP ACK (on 200 OK/SDP) message is routed on the Proxy Edge successfully (loose_route() deletes Route and handle_ruri_alias() sets $du). When GRUU is disabled - loose_route() for some reason replaces RURI (but lr=on ?) and sets $du. The configuration of the [WITHINDLG] route section is similar to the default script. IP: 1.1.1.1 is the Proxy Registrar IP: 2.2.2.2 is the Proxy Edge IP: 3.3.3.3 is the Phone (Phoner Lite) Successful case (GRUU enabled): 2022/01/21 20:08:20.745859 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:20.746952 2.2.2.2:5060 -> 3.3.3.3:5065 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK0a34.2b5076a5473cc42d883bf38969aa7dcf.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 3(11755) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: rr [loose.c:856]: after_loose(): Topmost route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa' is me 3(11755) DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header 3(11755) DEBUG: rr [loose.c:181]: find_next_route(): No next Route HF found 3(11755) DEBUG: rr [loose.c:886]: after_loose(): No next URI found 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: nathelper [nathelper.c:1202]: handle_ruri_alias(): setting dst_uri to <sip:3.3.3.3:5065> 3(11755) DEBUG: nathelper [nathelper.c:1224]: handle_ruri_alias(): rewriting r-uri to <sip:userid@2.2.2.2:5060 ;gr=009356C7-5A79-EC11-8468-538E88103B96> Unsuccessful case (GRUU disabled): 2022/01/21 20:08:47.851170 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:47.851963 2.2.2.2:5060 -> 2.2.2.2:5060 ACK sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK110e.af605388ece54d48387c1de53615d894.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 1(11753) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 1(11753) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 1(11753) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 1(11753) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 1(11753) DEBUG: rr [loose.c:676]: after_strict(): Next hop: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' is loose router 1(11753) DEBUG: rr [loose.c:726]: after_strict(): The last route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: rr [record.c:977]: add_rr_param(): rr_param_buf=<;did=3f8.6a83> 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: tm [t_lookup.c:1329]: t_newtran(): msg (0x7efe343f4998) id=11/11753 global id=10/11753 T start=0xffffffffffffffff 1(11753) DEBUG: tm [t_lookup.c:498]: t_lookup_request(): start searching: hash=57361, isACK=1 1(11753) DEBUG: tm [t_lookup.c:456]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0] 1(11753) DEBUG: tm [t_lookup.c:675]: t_lookup_request(): no transaction found 1(11753) DEBUG: tm [t_funcs.c:286]: t_relay_to(): forwarding ACK statelessly -- BR, Denys Pozniak

1 0

Experiences with Kamailio on Debian 11 Bullseye
by Daniel-Constantin Mierla 20 Jan '22

20 Jan '22

Hello, has anyone experienced issues with Kamailio running on Debian 11 Bullseye? It's now more than 2 months since Debian 11 release and I wonder if people started to migrate to it for production systems. So far I did only basic tests on it, although I have a Debian Testing that I used for compilation and simple tests during the past several months before Debian 11 release. If someone discovers issues or changes in behaviour with various libraries or applications/database servers, report them to the tracker to try to address them as soon as possible. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 08-11, 2021 (Europe Timezone) - Nov 22-25, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

8 14

Handling special characters - KEMI lua redis_ndb
by Muhammad Danish Moosa 20 Jan '22

20 Jan '22

Hi Guys, I am trying to do following code KSR.ndb_redis.redis_cmd("srvN", "SET var " .. testvar, "r"); it works fine but if testvar has special character like SPACE or % , it does not work. For example when testvar="asdsada%Dasda"; it starts considering %D as place holder and does not add key to redis (does not raise error as well) With Space I tried to use delimeters like "" or '' but none of them work. For now I have got it working through , search and replace but thats not a clean solution. Any suggestions? -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 1

Does cfgutils usleep block the process?
by mayamatakeshi 20 Jan '22

20 Jan '22

Hi, it seems execution of usleep eventually reaches this code: https://github.com/kamailio/kamailio/blob/master/src/core/ut.h#L508 so it seems it would block the process and so it might cause delay of packet processing if many processes do this at the same time. Can someone confirm this?

2 2

use $var or $avp in ds_list_exists & ds_select dispatcher function
by Patrick Karton 19 Jan '22

19 Jan '22

Hello im using kamailio 5.5.3 and im trying to use pseudo vars in ds_list_exists and ds_select function but it does not work. ds_list_exists return -2 with ds_list_exists("6") it works $var(dset)=6 or $avp(dset)=6 it does not work ds_list_exists("$var(dset)") --> return -2 ds_list_exists("$avp(dset)") --> return -2 Thanks.

1 0

Fwd: Possible memory leak on 5.5.x (new)?
by George Diamantopoulos 19 Jan '22

19 Jan '22

Forwarding my reply to the list, using gmail's reply button set Henning as the sole recipient :-\ ---------- Forwarded message --------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: Sat, 26 Jun 2021 at 02:25 Subject: Re: [SR-Users] Possible memory leak on 5.5.x (new)? To: Henning Westerholt <hw(a)skalatan.de> Hello Henning, Thanks for your reply. Here's what has come up after a few hours: shm55: https://pastebin.com/h9JCePmc shm54: https://pastebin.com/Nx5xEEnA It seems to me htable is the culprit? Are you seeing anything different? 54 has been running for 77020 seconds, 55 for 28521 (significantly less). I'm going to turn it off until we figure something out... BR, George On Fri, 25 Jun 2021 at 18:17, Henning Westerholt <hw(a)skalatan.de> wrote: > Hello, > > > > Good observation. Please run the memory statistics CLI commands to get > more hints about the module that might cause it (as per below link). Then > please report more details. If you can point to a particular module, you > can also open an issue on our tracker. > > > > https://www.kamailio.org/wiki/tutorials/troubleshooting/memory > > > > Cheers, > > > > Henning > > > > *From:* sr-users <sr-users-bounces(a)lists.kamailio.org> *On Behalf Of *George > Diamantopoulos > *Sent:* Friday, June 25, 2021 4:53 PM > *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > *Subject:* [SR-Users] Possible memory leak on 5.5.x (new)? > > > > Hello all, > > > > I'm still investigating the (most likely non-kamailio-related) memory leak > of my previous message to the list, there have been no developments so far. > I'll update if anything changes. > > > > This concerns a new finding which seems to affect kamailio 5.5.x. I have > two kamailio instances receiving the same traffic via round-robin. I > upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared > of any issues arising. I was lucky to do so, because with identical > configuration, 5.5.x seems to run out of SHM very quickly. Here are links > to graphs produced by our monitoring system: > > > > Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png > > New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png > > > > The configuration uses mtree, htable, vars and vns extensively. Has anyone > come across anything similar? Let me know if I can provide any further > information to help disect this. Thanks! > > > > BR, > > George >

2 8

Issues running mongodb command from kamailio
by Edo 18 Jan '22

18 Jan '22

Hi All, Please help! I am getting an error when trying to start kamailio after making some changes to the configuration file. Error is : *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 47-57: '('')' expected (function call)* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad expression* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad command* *atus=255/EXCEPTION* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 103: bad command* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 105: bad command* * used inside params of another function: xlog* *rse error in config file /etc/kamailio/kamailio.cfg, line 879, column 28: use of function execution inside params not allowed* *Server.* Changes made is : # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; #Start of change if (mongodb_find_one("mongodbsrv1", “telephone", “cXXX", " { \'did\': \'2XXXXXXXXXXXXXXXXX\' } ", "mgr1")) { xlog("response from mongodb is [[$mongodb(mgr1=>value)]]\n"); } #End of change append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } Please can someone tell me what I am doing wrong. Error says syntax but I have tried many permutations of the suggested syntax with no luck. Please help Thanks -- ----- Ekunwe EDO Network Services, Inc. Tel: 601.497.3932

2 4

REGISTER for MS Teams
by Lucian Romi 18 Jan '22

18 Jan '22

Hi, can register send from ms teams or I have to use Kamailio to send to my PBX.

2 1

JS Kemi TypeError: undefined not callable (property 'ifdef' of [object Object])
by Huseyin Kaan Koyukan 18 Jan '22

18 Jan '22

Hello Everyone, First of all, this is my first ever e-mail to the group so I'd like to thank all the developers who built this great product and thank everyone for the kind and helpful community. I've been using Kamailio for the past 8 months, I've had lots of fun with it. My issue is I'm trying to set-up JS Kemi interpreter for my Kamailio-Asterisk box. I translated the config file to JS kemi, which wasn't too bad but I'm getting a type error for the "KSR.kx.ifdef("WITH_ASTERISK")" statement that I used. Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:1478]: dukopen_KSR(): module 'KSR' has been initialized Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:473]: jsdt_sr_init_child(): loading js script file: /usr/local/etc/kamailio/kamailio-basic-kemi-jsdt.js Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: app_jsdt [app_jsdt_api.c:447]: jsdt_kemi_load_script(): failed running: TypeError: undefined not callable (property 'ifdef' of [object Object]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [core/sr_module.c:865]: init_mod_child(): error while initializing module app_jsdt (/usr/local/lib64/kamailio/modules/app_jsdt.so) (idx: 0 rank: -127 desc: [main]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [main.c:1671]: main_loop(): error in init_child(PROC_INT) -- exiting In syslog I can see that kex.so module is loaded successfully and 'KSR' has been initialized. I don't know what I'm doing wrong. A few sample JS Kemi code that I could find did not use the kx.ifdef function. Was this a new addition to the interpreter? Perhaps my Kamailio version (kamailio 5.5.0 x86_64/linux) does not support it? Thank you all in advance. Kaan.

1 0

Help with rewriting headers for NAT manually
by Chad 17 Jan '22

17 Jan '22

We are looking for some help (possibly a paid consultant) to help us with our Kamailio setup. To keep this as short as possible: we use Kamailio as a NAT proxy to bridge our external IP and our private IP asterisk servers (via dispatcher). However both the external IP and the internal IP that the Kamailio server uses are virtual IPs created by keepalived. Because of that neither mhomed nor fix_nated_contact work, and we use force_send_socket to direct the traffic. We run linux Debian 10 for the OS. Also we do not use a DB at all, everything is done with local config files. The problem is that when traffic goes out the Contact header has a private IP in it, like: Contact: <sip:##########@10.10.10.###]:5060> There are 2 possible solutions to this: 1. Make changes to linux, keepalived and/or Kamailio so that Kamailio recognize the virtual IPs so that mhomed and fix_nated_contact work as usual. 2. Create a manual header rewrite system. If solution #2: What we need to do is create a way to rewrite the contact header to the external IP on the way out, and on the way back rewrite it back to the internal server that the call is already connected to. Not sure if we will need to store those paths on the server or if we can do some kind of cheat with another persistant header like P-Preferred-Identity or P-Asserted-Identity (i.e. store the internal IP in the name field or something). If anyone out there know of a way to do this or wants to give it a try please reach out to me. Thank you all for your time. -- ^C Chad

5 27

Alternate to FoHSS for Kamailio IMS?
by Simon,Benjamin T 17 Jan '22

17 Jan '22

Hello, We are attempting to setup an instance of Kamailio to serve as an IMS for a test setup of 4G/5G NSA with Open5GS. The instructions on the Kamailio wiki and provided by Open5GS currently specify to use FoHSS as the HSS solution for Kamailio IMS. As this HSS solution has not seen significant development in a few years we are reticent to deploy it on our implementation. Is there an alternative open source HSS which can be deployed to serve as the HSS for Kamailio instead of FoHSS, and would there need to be any adjustments to Kamailio to reflect the deployment of this HSS instead of FoHSS? Thank you, Benjamin Simon Research Assistant Florida Institute for Cybersecurity Research, University of Florida Email: bsimon4(a)ufl.edu Disclaimer: Florida has a very broad public records law. As a result, any written communication created or received by the University of Florida will be made available to the public and media, upon request, unless otherwise exempt. Under Florida law, e-mail addresses are public records. If you do not want your e-mail address released in response to a public records request, do not send electronic mail to this address. Instead, contact by phone or in writing.

2 2

RR balance kamailio
by Gokan Atmaca 16 Jan '22

16 Jan '22

Hello I installed kamailio on Ubuntu. The version information is as follows. (0) I added two dispatcher sip servers. (1) When I enter the parameter as below in the configuration file , I get an error as follows. (2) I am getting RR module not available error. Packages installed are "kamailio-berkeley-modules, kamailio-tls-modules, kamailio-presence-modules" How can I solve the problem? (0) version: version: kamailio 5.5.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, 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: unknown compiled with gcc 9.3.0 (1) Dispatcher list: { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } DEST: { URI: sip:Y.Y.Y.Y FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } } } } } (2) if ( method=="INVITE" ) { ds_select_dst("1","4"); sl_send_reply("100","Trying"); forward(); exit(); } Err_log: Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

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Mid-dialog failover without virtual IP
by Muhammad Danish Moosa 14 Jan '22

14 Jan '22

Hi, I have a pair of kamailio stateful proxies and am trying to achieve Active-Passive failover model with DMQ dialog replication. Primary Proxy is working in the middle of 2 SBCs and all nodes are within the same network so NATing is NOT involved ;secondary proxy is accessible through VPN. It's pretty simple configuration, proxy is just adding and removing headers in the middle. So if I stop the kamailio service on primary IP , SBCs blacklist that and start sending new invites to secondary IP. That's all good. But if I reproduce this failover during a connected session , SBCs do not send mid-dialog requests/responses to failover IP. I can see they still try to send messages to primary SBC which obviously end up with timeout. I understand the most natural solution is HA with Virual IP but it's not possible because secondary IP is accessible through VPN. I tried to use DNS SRV record with priorities but that also does not work with mid-dialog messages, I also have tried to change record-route/via header to fqdn through advertise_address. Its not changing record-route but via only but I am still not getting expected results. Any guidance of possible solution(s) here will be appreciated. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

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Massive uploading of aliases through a dbtext
by Denys Pozniak 14 Jan '22

14 Jan '22

Hello! I need help uploading a list of aliases through a dbtext to use them via myself. It is likely that there is some configuration problem: *loadmodule "db_text.so"* *modparam("db_text", "db_mode", 1)modparam("db_text", "emptystring", 1)* *loadmodule "alias_db.so"modparam("alias_db", "db_url", "text:///etc/kamailio/dbtext")* *[root@kamailio]$ cat dbtext/dbaliasesid(int,auto) alias_username(string) alias_domain(string) username(string) domain(string)1::sip.test.local::* *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { }}* If done through a script (alias=sip.test.local), then everything works fine: *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { alias: { proto: * address: sip.test.local port: * } }}* -- BR, Denys Pozniak

2 4

Retrieve last via IP from SIP MEssage
by Adarsh Chauhan 14 Jan '22

14 Jan '22

Hi All, I have tried couple of methods to retrieve last via IP of SIP message but failed to retrieve it, trying to retrieve the source - @hf_value.via[-1].host, it simply prints the exact text back @hf_value.via[-1].host - $sel((a)hf_value.via[-1]) it doest give the IP but the entire via value SIP/2.0/TCP 172.31.30.144;branch=z9hG4bKf8bd.753341d2000000000000000000000000.0.host @via[-1].host it also simply prints the exact text back @via[-1].host Regards, Adarsh Chauha

2 1

EVAPI workload allocation across processes
by Alex Balashov 13 Jan '22

13 Jan '22

Hi, Given several EVAPI worker processes: modparam("evapi", "workers", 4) # kamcmd -s /run/kamailio/kamailio_ctl ps | fgrep -i evapi 539257 EvAPI Dispatcher 539258 EvAPI Worker 539259 EvAPI Worker 539260 EvAPI Worker 539261 EvAPI Worker How should one understand the way that the EVAPI workload is allocated among them? My best guess is that the EvAPI dispatcher process is the sole accept()or of client connections and the place where all relaying to client connections is done, regardless of their number, and that the ‘workers’ represent an async pool of processes in which TM transactions can be resumed or other work can be done when an outside caller sends a message to the dispatcher. However, the documentation[1] is a little vague on the meaning of the ‘workers’ modparam: "Number of worker processes to be started to handle incoming messages from remote applications.” Or maybe it’s not. Just wanted to confirm that my understanding is correct. Thanks! — Alex [1] https://kamailio.org/docs/modules/5.5.x/modules/evapi.html#evapi.p.workers -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Reverse Kamailio Dispatcher order
by Adarsh Chauhan 12 Jan '22

12 Jan '22

Hi All, I was looking for a way where i could reverse the order of priority set on destinations in dispatcher file. for eg: if i have the following file: 1 sip:10.252.168.41:5060;transport=tcp 0 5 # (US) 1 sip:10.252.168.42:5060;transport=tcp 0 4 # (US) 1 sip:10.252.168.43:5060;transport=tcp 0 3 # (EU) 1 sip:10.252.168.44:5060;transport=tcp 0 2 # (EU) I wanted to use top to bottom when in US region and bottom to top when in EU region. One option is to make another set, however that will force Kamailio to send ping (OPTIONS) twice when probing. Regards, Adarsh Chauhan

4 4

Suggestion on receiving AAR
by Jyoti Bansal 10 Jan '22

10 Jan '22

Hello all, We are trying to connect kamailio ims to a pcrf, connection is successful. Whenever we register through bhoghe client , we receive AAR on pcrf from kamailio ims(pcscf), so just need a suggestion can we disable that till we receive invite from the soft phone? Can anyone help on this? Thanks, Jyoti bansal -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

Call Preservation Mode in Kamailio
by varun pratapsingh 10 Jan '22

10 Jan '22

Happy New Year 2022 Everyone. Wish you a lot of innovation this year! I was just curious to know if any of us had handled the mid-call failures using the Kamailio. I am writing a SIP Client to handle the failures by using the DNS SRV based failover and Fallback approach where I am using two Kamailio instances as Outbound Proxy resolved by DNS SRV lookup. Everything is fine except for on-going calls. How to handle the mid-call failovers? Is there any way in Kamailio like Call Preservation mode or something? Any suggestions or ideas to handle midcall failover so that if a current Outbound Proxy goes down then all the ongoing calls can be seamlessly routed through another Kamailio (secondary). Thanks and Regards Varun

3 2

Kamailio call flows with Asterisk
by Nauman Sulaiman (SESSIONTALK) 05 Jan '22

05 Jan '22

Hi, We are using Asterisk as a PBX with users directly registered to Asterisk and Asterisk registering to SIP trunks. We are now looking to put Kamailio in front of Asterisk to handle SIP registrations from the SIP clients. In a ‘typical’ architecture should we keep the SIP trunk registrations on Asterisk or is Kamailio used for this? We want to keep Asterisk as the RTP proxy so we don’t want a call setup by Kamailio with RTP then going direct between user agents. Regards Nauman

3 2

Calling t_relay([host, port]) with variables
by Nick J 05 Jan '22

05 Jan '22

Hi all, I've been banging my head on the table with this one and hoping someone has some insight here. I'm trying to use a variable as the host & port input in t_relay() - t_relay(host, port) This works fine if statically set: t_relay("10.0.1.1", "5060"); But if I change the host or the port to a variable like this: $avp(destip) = "10.0.1.1"; $avp(dstport) = "5060"; t_relay($avp(destip), $avp(dstport)); Then Kamailio fails to start, calling out the t_relay line: kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio-routelogic.cfg, line 62, column 35: bad command I get the same behavior with pseudovariables and vars as well: $var(destip) = "10.0.1.1"; $var(dstport) = "5060"; t_relay($var(destip), $var(dstport)) Any tips or pointers would be much appreciated. Thanks! Nick

4 3

Features of Kamailio as SBC
by Nattha Vudhiprasittipol 04 Jan '22

04 Jan '22

Hello I would like to know all the feature lists of Kamailio as SBC. Could you please send me the document? I cannot find it on the website. Best regards Nattha Vudhiprasittipol Product Development Engineer Creaturelab Network Solution Co.,Ltd 357 Charansanitwong Rd., Bang-Bamru, Bangphlat, Bangkok 10700, THAILAND Email: nattha.v(a)creaturelab.co.th Website: www.creaturelab.co.th

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Recommendation for compatible IMS soft phones
by Ashay Matekar 04 Jan '22

04 Jan '22

Hi Team, We were trying to test a call flow with Kamailio IMS that requires preconditio. Do you know any free or paid IMS softphones that support preconditio and are compatible with Kamailio IMS. -- *Thanks and Regards,* *Ashay Matekar* Associate Manager Engineering | +91 932446823 Great Software Laboratory | www.gslab.com LinkedIn <https://www.linkedin.com/company/gs-lab> | Twitter <https://twitter.com/_gslab> | Email Disclaimer <https://www.gslab.com/privacy-policy/#email-disclaimer> -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

Catching Register 200 OK, then using it to validate other packets.
by Richard Edmands 04 Jan '22

04 Jan '22

Hey Kamailio Users, I've written a TLS-UDP gateway. It works well, but I want to reduce the internet noise it pushes through. I have an idea on what I'd like to try. Wondering if it would be possible to catch the Register 200 OK, store it and then when receiving packets it future to then validate them against the data? I've managed to capture a 200 OK, but the save("location") function says it will return it's own 200 OK which isn't the plan since we just want to capture the location. Anyone done something similar and got some working example code? Is this a bad idea for some reason?

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