- sr-users - kamailio.org

Kamailio in docker container with TLS enabled using Let's Encrypt

by Giovanni Tommasini

Hi guys, starting from Fred article https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ we move step forward creating this small PoC to deploy Kamailio in docker container with TLS enabled using Let's Encrypt https://github.com/evoseed/kamailio-tls-letsencrypt What do you think? Any feedback? Any mistakes to correct or improvements to make? Thanks a lot Have a good evening Giovanni Tommasini *evoseed s.r.l.* phone: +39 340 11 28 800 mail: giovanni.tommasini(a)evoseed.io site: evoseed.io address: Via Lucrezio 13, Trieste Italy

2 years, 6 months

2
2
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sanity from uri check

by Juha Heinanen

I have in config: modparam("sanity", "uri_checks", 3) /* RURI, From */ and then config call sanity_check() on incoming requests: if (!sanity_check()) { xnotice("Request from <$var(src_ip)> has invalid syntax\n"); exit; }; When incoming request has: From: sipp <sip:@192.168.255.95:5062>;tag=1 no notice is printed to syslog. >From RFC 3261: SIP-URI = "sip:" [ userinfo ] hostport uri-parameters [ headers ] userinfo = ( user / telephone-subscriber ) [ ":" password ] "@" user = 1*( unreserved / escaped / user-unreserved ) My understanding of that is that there needs to be at least one character before @ sign. -- Juha

2 years, 6 months

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properly removing First Via header received

by Patrick Karton

i want to remove the First Via Header received from uac to kamailio. so in my script i use remove_hf("Via") but in the relayed request i see that ";received=x.x.x.x;rport=xxxx" is left without any header before it. i think internally kamailio adds a lump before i call the remove_hf function. how to do it properly ? Thanks

2 years, 6 months

2
2
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Forking for scripts?

by Matthias Urlichs

Hello, Is it possible to run a background Python script within Kamailio? I.e. call "fork_process" and then a non-terminating "python_exec()" from the child? -- -- regards -- -- Matthias Urlichs

2 years, 6 months

2
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Question about a future 5.6 release

by Stefan-Cristian Mititelu

Hi, I want to ask you if there is an approximate timeline for a kamailio 5.6 release? We are planning an upgrade on our systems and are weighing different options. Thank you Stefan

2 years, 6 months

2
1
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typo in documentation

by David Villasmil

Hello all, I was trying to implement db_text and the documentation states it should be something like modparam("usrloc|auth_db", "db_url", "db_text:///tmp/kamailiodb") but that doesn't work, it's modparam("usrloc|auth_db", "db_url", "text:///tmp/kamailiodb") so this line should probably be updated: https://github.com/kamailio/kamailio/blob/master/src/modules/db_text/doc/db… Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 years, 6 months

3
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load to htable from flat file

by David Villasmil

Hello guys, Is this possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 years, 6 months

5
5
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RTPEngine inject DTMF

by Bendersky

Hello Everyone, I am trying inject DTMF and getting error message. ERROR:rtpengine:rtpe_function_call: proxy replied with error: No SSRC context present for DTMF injection I have full call rtpengine debug which I can provide upon request. Rtpengine version rtpengine-9.5.4.1-9.fc32.x86_64 Volga629

2 years, 6 months

1
0
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clean an easy way to get sip uri of contact header

by Patrick Karton

hello, i want a generic way to get the sip uri in contact header. for example if Contact: Sipp <sip:6565@ip;ov=65;gid=45>;tid=700 i want a way to get both <sip:6565@ip;ov=65;gid=45> (without header param) and <sip:6565@ip;ov=65;gid=45>;tid=700 (with header param) Thanks.

2 years, 6 months

3
3
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limit the length of Via branch parameter

by Patrick Karton

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 years, 6 months

2
3
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Table : address - address_reload (exported MI function) - expect to call this function in kamailio.cfg to reload in cache the address

by Youssef Boujraf

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 years, 6 months

2
1
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WebSocket ping is not being sent

by Xuo Guoto

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.

2 years, 6 months

3
5
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Any clever ways to strip down a user portion of an RURI?

by Brooks Bridges

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.

2 years, 6 months

3
4
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Interconnect with Zoom

by Matthias Urlichs

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

2 years, 6 months

3
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Kamailio + RADCLI

by Marrold

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 years, 6 months

2
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Kamailio 5.5 TLS critical log in Debian 11

by mayamatakeshi

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?

2 years, 6 months

3
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add a=fmtp: if it does not exist in sdp

by Krzysztof Drewicz

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

2 years, 6 months

3
3
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Diversion Header & Header management - $hdrc

by Brandon Armstead

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 years, 7 months

2
1
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Kamailio 5.4.7 crash

by Igor Olhovskiy

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

2 years, 7 months

1
0
0 0

Kamailio v5.4.8 Released

by Daniel-Constantin Mierla

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/

2 years, 7 months

1
0
0 0

Releasing Kamailio v.5.4.8

by Daniel-Constantin Mierla

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 years, 7 months

2
3
0 0

rtpengine - SRTP <> RTP missing a=crypto

by Rhys Hanrahan

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.

2 years, 7 months

4
13
0 0

Kamailio + KEMI + topos + change_reply_status

by Marrold

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 years, 7 months

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kamailio.cfg : behind the NAT (listen and advertise) understanding

by Youssef Boujraf

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

2 years, 7 months

3
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simple GUI for Kamailio as a SBC - could be paid

by Krzysztof Drewicz

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/>

2 years, 7 months

5
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OS-assigned TLS source Port#?

by Matthias Urlichs

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 years, 7 months

2
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ndb_redis - redis-cluster error

by Maharaja Azhagiah

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 years, 7 months

2
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Problem with auth_ephemeral and parse_uri(): bad port in uri

by Vlasis Chatzistavrou

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?

2 years, 7 months

4
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t_on_failure => what variables being reset to which values?

by Benoît Panizzon

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 years, 7 months

2
3
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KEMI jsdt CommonJS-based module

by Ian Carlson

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 years, 7 months

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RTP Engine early media issue one to many

by Filippo Graziola

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

2 years, 7 months

1
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ACK and DNS failover

by Julien Klingenmeyer

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

2 years, 7 months

3
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kamailio and certificate renewal

by Aymeric Moizard

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 years, 7 months

2
1
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Re-using an existing TLS connection

by Matthias Urlichs

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

2 years, 7 months

3
3
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Info: SIP resources collection

by Daniel-Constantin Mierla

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/

2 years, 7 months

4
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kamailio : how to xlog the value of "myself" ?

by Youssef Boujraf

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

2 years, 7 months

3
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dispatcher.reload

by David Villasmil

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

2 years, 7 months

4
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Dispatcher Load Balance with SIP Trunk

by warawich

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 years, 7 months

2
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SQLOPS sql_query_async() not functioning in an async manner?

by Ben Kaufman

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.

2 years, 7 months

4
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kamailio :?==?utf-8?q? ROUTE[AUTH] - IPAUTH - use case

by Youssef Boujraf

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

2 years, 7 months

1
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Issue with Authenticating outbound Call

by Victor ku

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 years, 7 months

2
2
0 0

reinvite with 2 video lines in SDP

by Filippo Graziola

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 years, 7 months

2
2
0 0

Enable switch redirect to other SIP server source

by bayu p

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 years, 7 months

2
1
0 0

kamailio success story

by Valentin Christoph

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 years, 7 months

2
1
0 0

SIP server port service cannot be changed from default value (5060)

by bayu p

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

2 years, 7 months

4
9
0 0

Kamailio v5.5.4 Released

by Daniel-Constantin Mierla

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 years, 7 months

2
2
0 0

Releasing Kamailio v5.5.4

by Daniel-Constantin Mierla

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/

2 years, 7 months

1
1
0 0

Info: sipexer v1.0.0 - sip cli tool

by Daniel-Constantin Mierla

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

2 years, 7 months

10
12
0 0

allow_source_address_group in failure route

by Marrold

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 years, 7 months

2
1
0 0

Missing contact in 180 ringing causes wrong routing

by Xuo Guoto

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.

2 years, 7 months

1
0
0 0

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