- sr-users - kamailio.org

t_check_trans in Lua
by nizam＠ctapps.com 06 Sep '24

06 Sep '24

Dear experts, Would appreciate if anyone can explain why in the given sample Lua script the 'Handle Retransmission' block checks the return code of t_check_trans (if KSR.tm.t_check_trans == 0 then), but in the Kamailio native config standard file if just calls the t_check_trans() function without checking the return code. What is the intention here and how does it work? Thank you in advance. Handle retransmission script from native config file: if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } Handle retransmission from sample Lua script: if not KSR.is_ACK() then if KSR.tmx.t_precheck_trans() > 0 then KSR.tm.t_check_trans(); return 1; end if KSR.tm.t_check_trans() == 0 then return 1; end end

2 2

secsipid and secsipid_proc module RPM build
by Володимир Іванець 04 Sep '24

04 Sep '24

Hello! I was wondering if someone was trying to build Kamailio RPM packages containing modules secsipid and secsipid_proc and faced the same problem. I compiled secsipidx library and copied secsipid.h, libsecsipid.h as well as libsecsipid.a files into the source directory of secsipid_proc module. And when I'm trying to build RPM packages, I get the following error: CC (gcc) [M secsipid_proc.so] secsipid_proc_mod.o secsipid_proc_mod.c:28:10: fatal error: secsipid.h: No such file or directory #include <secsipid.h> ^~~~~~~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: secsipid_proc_mod.o] Error 1 make[1]: *** [Makefile:508: modules] Error 1 make[1]: Leaving directory '/builddir/build/BUILD/kamailio-5.8.2/src' make: *** [Makefile:34: every-module] Error 2 Thanks a lot!

2 4

Kamailio v5.8.3 Released
by Daniel-Constantin Mierla 04 Sep '24

04 Sep '24

Hello, Kamailio SIP Server v5.8.3 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.3. For more details about version 5.8.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/09/kamailio-v5-8-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, October 8-10, 2024 -- asipto.com

1 0

Permissions module - automatic reload
by Sergio Charrua 04 Sep '24

04 Sep '24

Hi all, Just need some clarifications about the Permissions module: I currently use it with db_mode = 1 and was expecting that the parameter "trusted_cleanup_interval" would refresh the cache tables every X seconds, but apparently this is not the case. I was looking for a way for Kamailio to automatically reload the address table into memory every 5 minutes, for example, and I (mis)understood that "trusted_cleanup_interval" would do the trick, but it doesn't work as expected. Question: is there any command, besides kamcmd, to automatically reload the permissions table? I would like to avoid having to add a line in the crontab.... Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

1 0

Using a different OpenSSL
by David Cunningham 04 Sep '24

04 Sep '24

Hello, We have compiled openssl 3.0.9 from source because it's FIPS validated, and want to use it with Kamailio. The server also has the Ubuntu openssl 3.0.2 package installed. Does anyone know how we can tell Kamailio to use the openssl library in /opt/openssl/lib64, and how we can verify that it really is using it? Thanking you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

5 13

Releasing v5.8.3
by Daniel-Constantin Mierla 04 Sep '24

04 Sep '24

Hello, I am considering to release Kamailio v5.8.3 (out of branch 5.8) sometime next week, most likely on Wednesday, Sep 4, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

kamailio stri shaken configuration
by Vishal Pai 02 Sep '24

02 Sep '24

Hello All I need to integrate the kamailio with stri shaken with pem. What is the best way to achieve this? Thanks

3 2

app_mono module to be archived
by Victor Seva 31 Aug '24

31 Aug '24

Hi kamilians, we have the intention to archive the app_mono module in a few[0]. Most likely at the end of next week. If you use app_mono and you don't want to see it fade away now it's the time to rise your voice. Cheers, Victor [0] https://github.com/kamailio/kamailio/pull/3964

1 0

How to restrict Kamailio adding Local VIA header
by panda.paresh＠gmail.com 30 Aug '24

30 Aug '24

Hi All, I am trying to replay a request to the destination via Kamailio, and I am using t_relay_to_tcp for that , but in that case Kamailio is adding its own address as the top VIA header. Is there a way I can instruct Kamailio not to add local via header ?

3 4

http_async and tm
by alexis 30 Aug '24

30 Aug '24

Hello all, context first, we have an REST API that performs queries to external devices in the network (diameter to DRA's, REST to different servers) and based on n conditions returns the content for a Contact header to be used in a SIP 302. Now we're consuming this API with http_client (synchronously) and as there's no way to speed up the API (pipeline executions, delays on external api's etc etc) we're hitting a limit where all children become busy waiting for the API to answer. So i decided to move to http_async_client and started working on it on the lab with this first and base concept to test. request_route { #for testing purposes only if(is_method("ACK")){ exit; } $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = 500; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string", "event", "sip-routing", "$var(cre_query)"); xlog("L_INFO","API ASYNC ROUTING REQUEST: $var(cre_query)\n"); $http_req(body) = $var(cre_query); t_newtran(); http_async_query("http://192.168.86.128:8000/", "CRE_RESPONSE"); } http://192.168.86.128:8000/ receives the POST, randomly creates a delay between 0.5 and 1 second and responds (simulating the real api with an excess delay to probe the concept) Then route[CRE_RESPONSE] { if ($http_ok && $http_rs == 200) { xlog("L_INFO","CRE RESPONSE: $http_rb\n"); # for testing purpose, Contact content will be replaced from the received api response append_to_reply("Contact: <sip:1234@google.com>\r\n"); send_reply(302,"Moved Temporarily"); exit; } send_reply(500, "Internal error"); exit; } INVITE is received and processed, API is called, after API response, 302 is replied and then an ACK (ignored by now). Situation is that the 302 retransmitted 37 1519.846253067 192.168.86.34 → 192.168.86.128 SIP/SDP 585 Request: INVITE sip:service@192.168.86.128:5060 | 38 1519.848100380 192.168.86.128 → 192.168.86.34 SIP 318 Status: 100 Trying | 39 1520.094997642 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 40 1520.102323728 192.168.86.34 → 192.168.86.128 SIP 453 Request: ACK sip:service@192.168.86.128:5060 | 41 1520.591300933 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 42 1521.591061065 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 43 1523.591227956 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [timer.c:634]: wait_handler(): finished transaction: 0x7f6d75184cc8 (p:0x7f6d74f600c8/n:0x7f6d74f600c8) 18(24) DEBUG: tm [h_table.c:132]: free_cell_helper(): freeing transaction 0x7f6d75184cc8 from timer.c:643 Any help to avoid the retransmission and make the transaction just finish right after the 302 will be appreciated. regards

6 27

apply keepalive to only udp
by James Lipski 30 Aug '24

30 Aug '24

Hello, I wanted to know if this can be accomplished, or if I'm doing something wrong. What I'm hoping to accomplish is to have kamailio periodically send OPTIONS packets to endpoints that are connected over UDP only. As of right now, I currently have the following configured under usrloc: modparam("usrloc", "timer_interval", 5) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 0) modparam("usrloc", "ka_mode", 1) modparam("usrloc", "ka_method", "OPTIONS") modparam("usrloc", "ka_from", "sip:ping@sip-domain.com") modparam("usrloc", "ka_domain", "sip-domain.com") modparam("usrloc", "ka_timeout", 125) modparam("usrloc", "ka_interval", 60) With these parameters, all endpoints will be sent OPTIONS packets periodically as expected. When setting "ka_mode" to "4", as expected, endpoints registered with UDP only receive OPTIONS packets; which is what I wanted. The issue that I'm having with these parameters is that endpoints registered with anything other that UDP are triggering a contact-expired event after a little over 2 minutes. I have the registrar set to 3600 as an expiry; I'm assuming the contact is expiring due to the "ka_timeout" value. -------------------------- Is there a way to only have keep-alive packets/ and timeouts apply to endpoints registered on UDP? Thank you.

3 5

Kamailio integration with REST services
by Sergio Charrua 29 Aug '24

29 Aug '24

Hi all! I need to integrate a REST Service with Kamailio. Integration is easy, not a problem, but my concern is that the REST Services has a JWT token authentication required for each request, common to many (it not all) REST Services. So, it is required to POST for authentication (username and password) and once I get authentication done, and receive a JWT token, I will need to use this token in all other requests to the REST Service. The token is valid for 24h. My question is: - how to save the token for use during 24h, on all calls/sessions, without the need to authenticate on every call? - how to detect the token has expired and re-authenticate? I looked for some modules, but could not find any that suited me. I am thinking of executing a Python script for this, but concerned about latency and PDD.... Any thoughts? Any suggestions? Any known best-practice? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

6 12

kamailio randomly crashes
by Maharaja Azhagiah 28 Aug '24

28 Aug '24

Hi all, I have kamailio running in GKE (Google Kubernetes) and recently seeing kamailio crashes frequently and I couldn't find the reason for the crash kamailio version version: kamailio 5.4.9 (x86_64/linux) 7d022d 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, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 7d022d compiled on 14:26:30 Jul 10 2024 with gcc 8.3.0 kamailio core generation: Core was generated by `kamailio -e -E -DD -M 128 -m 2048'. Program terminated with signal SIGSEGV, Segmentation fault. #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 416 if (rd->carriers[i]->domains[j]->sum_prob == 0.0) { bt: (gdb) bt #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 #1 0x00007ca474c8cc50 in reload_route_data () at cr_data.c:178 #2 0x00007ca474cabfdc in cr_rpc_reload_routes (rpc=0x7ca4f5a32aa0 <func_param>, c=0x7ca4f5a329c0 <_jsonrpc_ctx_global>) at cr_rpc.c:59 #3 0x00007ca4f5a158a1 in jsonrpc_exec_ex (cmd=0x7ffd774f34b0, rpath=0x0) at jsonrpcs_mod.c:1425 #4 0x00007ca4f5a15ccd in jsonrpc_exec (msg=0x7ca4f641a9f8, cmd=0x7ca4f61bde20 "x\335\033\366\244|", s2=0x0) at jsonrpcs_mod.c:1445 #5 0x00005745eb3fcb19 in do_action (h=0x7ffd774f3dc0, a=0x7ca4f61f1a50, msg=0x7ca4f641a9f8) at core/action.c:1085 #6 0x00005745eb40b35c in run_actions (h=0x7ffd774f3dc0, a=0x7ca4f61f0468, msg=0x7ca4f641a9f8) at core/action.c:1584 #7 0x00007ca474b92f8d in timer_handler (ticks=96802941, tl=0x7ca47b2f7950, data=0x7ca47b2f7868) at timer.c:216 #8 0x00005745eb49e3c4 in slow_timer_main () at core/timer.c:1105 #9 0x00005745eb3a85bd in main_loop () at main.c:1765 #10 0x00005745eb3b34c3 in main (argc=8, argv=0x7ffd774f4688) at main.c:2942 bt full: (gdb) bt full #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 res = 0x7ca4f636a4a8 row = 0x7ca4f642a7f8 i = 0 j = 110 ret = 56 tmp_carrier_data = 0x7ca479ff5068 query_str = {s = 0x7ca474cdd800 <query> "SELECT DISTINCT domain FROM carrierroute WHERE carrier=8", len = 56} tmp_scan_prefix = {s = 0x5745ed889f68 "13235791374", len = 11} tmp_rewrite_host = {s = 0x5745ed889f7a "sip.telnyx.com;transport=TLS", len = 28} tmp_rewrite_prefix = {s = 0x5745ed889f99 "", len = 0} tmp_rewrite_suffix = {s = 0x5745ed889f9a "", len = 0} tmp_host_name = {s = 0x40 <error: Cannot access memory at address 0x40>, len = 1965559808} tmp_reply_code = {s = 0x7ca474ccf2b0 <__func__.5843> "reload_route_data", len = 1959587872} tmp_comment = {s = 0x5745ed889f9b "IMPORTED - WTJHSX", len = 18} p_tmp_comment = 0x7ffd774f30b0 __func__ = "load_route_data_db" n = 0 #1 0x00007ca474c8cc50 in reload_route_data () at cr_data.c:178 old_data = 0x5745ed850150 new_data = 0x7ca482a48840 i = 22341 __func__ = "reload_route_data" #2 0x00007ca474cabfdc in cr_rpc_reload_routes (rpc=0x7ca4f5a32aa0 <func_param>, c=0x7ca4f5a329c0 <_jsonrpc_ctx_global>) at cr_rpc.c:59 __func__ = "cr_rpc_reload_routes" #3 0x00007ca4f5a158a1 in jsonrpc_exec_ex (cmd=0x7ffd774f34b0, rpath=0x0) at jsonrpcs_mod.c:1425 rpce = 0x7ca4f6425fb8 ctx = 0x7ca4f5a329c0 <_jsonrpc_ctx_global> ret = -1 nj = 0x0 val = {s = 0x5745ed845010 "cr.reload_routes", len = 16} scmd = {s = 0x7ca4f61bdd78 "{\"jsonrpc\": \"2.0\", \"method\": \"cr.reload_routes\", \"id\": 2}", len = 57} __func__ = "jsonrpc_exec_ex" #4 0x00007ca4f5a15ccd in jsonrpc_exec (msg=0x7ca4f641a9f8, cmd=0x7ca4f61bde20 "x\335\033\366\244|", s2=0x0) at jsonrpcs_mod.c:1445 scmd = {s = 0x7ca4f61bdd78 "{\"jsonrpc\": \"2.0\", \"method\": \"cr.reload_routes\", \"id\": 2}", len = 57} __func__ = "jsonrpc_exec" #5 0x00005745eb3fcb19 in do_action (h=0x7ffd774f3dc0, a=0x7ca4f61f1a50, msg=0x7ca4f641a9f8) at core/action.c:1085 ret = -5 v = 0 dst = {send_sock = 0x0, to = {s = {sa_family = 41744, sa_data = "6\366\244|\000\000\035u\271t\244|\000"}, sin = {sin_family = 41744, sin_port = 63030, sin_addr = {s_addr = 31908}, sin_zero = "\035u\271t\244|\000"}, sin6 = {sin6_family = 41744, sin6_port = 63030, sin6_flowinfo = 31908, sin6_addr = {__in6_u = {__u6_addr8 = "\035u\271t\244|\000\000\005u\271t\244|\000", __u6_addr16 = {29981, 29881, 31908, 0, 29957, 29881, 31908, 0}, __u6_addr32 = {1958311197, 31908, 1958311173, 31908}}}, sin6_scope_id = 2001680800}, sas = {ss_family = 41744, __ss_padding = "6\366\244|\000\000\035u\271t\244|\000\000\005u\271t\244|\000\000\240\071Ow\375\177\000\000/.:\365\244|\000\000\320:Ow\375\177\000\000\327\337j\353EW\000\000\320\071Ow\375\177\000\000\020@ \016\366\244|\000\000\030\245\066\366\244|\000\000\020@\016\366\244|\000\000\020:Ow\375\177\000\000\356u\236\365\244|\000\000\030\245\066\366\244|\000\000\000\000\000\000\000\000\000", __ss_align = 134217728}}, id = 130848392, send_flags = {f = 0, blst_imask = 0}, proto = 56 '8', proto_pad0 = 5 '\005', proto_pad1 = 36} tmp = 0x7ca4f641a9f8 "\344\032" new_uri = 0xed3 <error: Cannot access memory at address 0xed3> end = 0x7ca474b9751b "\r\n" crt = 0x2 <error: Cannot access memory at address 0x2> cmd = 0x7ca4f617b7d8 len = 0 user = 15 uri = {user = {s = 0x9774f3800 <error: Cannot access memory at address 0x9774f3800>, len = 621}, passwd = {s = 0x400000000 <error: Cannot access memory at address 0x400000000>, len = 1721781302}, host = {s = 0x0, len = -2147483648}, port = {s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, params = {s = 0x7ffd774f38e0 "09Ow\375\177", len = -167557169}, sip_params = { s = 0x7ca4f53704f0 <dialog_table_name> "l^5\365\244|", len = -164192016}, headers = {s = 0x1700000000 <error: Cannot access memory at address 0x1700000000>, len = 2001681040}, port_no = 14800, proto = 30543, type = 32765, flags = (unknown: 4130775280), transport = {s = 0x0, len = 2009201968}, ttl = {s = 0x6774f3a50 <error: Cannot access memory at address 0x6774f3a50>, len = -181058744}, user_param = { s = 0x17200000000 <error: Cannot access memory at address 0x17200000000>, len = 2099608304}, maddr = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -344235896}, method = {s = 0x0, len = -2147483648}, lr = { --Type <RET> for more, q to quit, c to continue without paging-- s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, r2 = {s = 0x7ffd774f38d0 "`9Ow\375\177", len = -166931844}, gr = {s = 0x7ca4f53704f0 <dialog_table_name> "l^5\365\244|", len = -164192016}, transport_val = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -181636538}, ttl_val = {s = 0x7ffd774f3930 "`9Ow\375\177", len = -167555326}, user_param_val = {s = 0x0, len = 2}, maddr_val = { s = 0x7ca4f60c1f77 <db_mysql_submit_query> "UH\211\345AWAVAUATSH\201\354\210", len = -166889584}, method_val = {s = 0x177d6038c8 <error: Cannot access memory at address 0x177d6038c8>, len = 2001681040}, lr_val = {s = 0xe00000000 <error: Cannot access memory at address 0xe00000000>, len = 1958311183}, r2_val = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -164191464}, gr_val = { s = 0x7ffd774f39f0 "\373\377\377\377", len = -344973614}} next_hop = {user = {s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, passwd = {s = 0x7ffd774f3720 "\020\070Ow\375\177", len = -166975523}, host = { s = 0x7ffd774f3740 "\020\070Ow\375\177", len = -166887566}, port = {s = 0x7ca4f60431d0 <sql_str> "@@w\355EW", len = -164192016}, params = {s = 0x7ffd774f3d50 "", len = -164192016}, sip_params = {s = 0x0, len = -30810208}, headers = {s = 0x0, len = -181051830}, port_no = 8055, proto = 62988, type = 31908, flags = (unknown: 4128077712), transport = { s = 0x4774f37d0 <error: Cannot access memory at address 0x4774f37d0>, len = 2001680544}, ttl = {s = 0x669fa379 <error: Cannot access memory at address 0x669fa379>, len = -164192016}, user_param = { s = 0x5745ed7743b0 "'sip:+13194676789@us.test.com',NULL,0)", len = 1721781302}, maddr = {s = 0x400000000 <error: Cannot access memory at address 0x400000000>, len = -167513741}, method = { s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, lr = {s = 0x7ffd774f3810 "\340\070Ow\375\177", len = -167567737}, r2 = { s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, gr = {s = 0x7ffd774f3810 "\340\070Ow\375\177", len = -166977673}, transport_val = {s = 0x7ca4f60431d0 <sql_str> "@@w\355EW", len = -164192016}, ttl_val = {s = 0x7ffd774f3a90 "\330\267\027\366\244|", len = 23}, user_param_val = { s = 0x5745ed77417a "3071,3795,'6e1a9c34-cec5-4ebb-bcd9-3356d80bcf9e',' sip:+13194004696@sip.telnyx.com','c5SS3F27B8rDN',' sip:+13194676789@us.test.com','f9b9b2c48ef41ef90eb05056b0a794','tls: 10.16.0.74:520"..., len = -164192016}, maddr_val = {s = 0x7ca4f60dcf18 "`", len = -31060352}, method_val = {s = 0x0, len = -180627954}, lr_val = {s = 0x7ca4f60c1f77 <db_mysql_submit_query> "UH\211\345AWAVAUATSH\201\354\210", len = -166889584}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} u = 0x7ca4f5817bc8 port = 0 dst_host = 0x7ca474b9751d i = -344941541 flags = 0 avp = 0x7ffd774f3c40 st = {flags = 3984015424, id = 22341, name = {n = -310010607, s = {s = 0x5745ed859d11 "", len = 2001679968}, re = 0x5745ed859d11}, avp = 0x66a04c36} sct = 0x60f60e6120 sjt = 0x5f60e6a50 rve = 0x8000000 mct = 0x0 rv = 0xffffffff rv1 = 0x5745eb7fc993 c1 = {cache_type = 10, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 0}, re = 0x0}, pval = {rs = {s = 0x0, len = 0}, ri = 2001679912, flags = 32765}}, i2s = "`6Ow\375\177\000\000\031\241\a\366\244|\000\000\250\003\000\000\000"} s = {s = 0xffffffff <error: Cannot access memory at address 0xffffffff>, len = -164179120} srevp = {0x3, 0x64870454268b4200} evp = {data = 0x0, obuf = {s = 0x0, len = 0}, rcv = 0x0, dst = 0x0, req = 0x0, rpl = 0x0, rplcode = 0, mode = 0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = { number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __func__ = "do_action" #6 0x00005745eb40b35c in run_actions (h=0x7ffd774f3dc0, a=0x7ca4f61f0468, msg=0x7ca4f641a9f8) at core/action.c:1584 t = 0x7ca4f61f1a50 ret = 1 tvb = {tv_sec = 0, tv_usec = 0} tve = {tv_sec = 0, tv_usec = 0} tz = {tz_minuteswest = 60, tz_dsttime = 0} tdiff = 4131498432 __func__ = "run_actions" #7 0x00007ca474b92f8d in timer_handler (ticks=96802941, tl=0x7ca47b2f7950, data=0x7ca47b2f7868) at timer.c:216 msg = 0x7ca4f641a9f8 a = 0x7ca47b2f7868 ra_ctx = {rec_lev = 1, run_flags = 0, last_retcode = 1, jmp_env = {{__jmpbuf = {95957815091336, 8333654393230933038, 0, 2147483648, 134217728, 95957815380371, 8333654393012829230, 2510245087605373998}, __mask_was_saved = 0, __saved_mask = {__val = {137047930071236, 0, 25985340079663656, 140726605135616, 137045787154760, 70721158768, 137045787160648, 140726605135472, 137045776334895, 1095350878208, 137045782276504, 140735195070144, 140726605135504, 137045776144708, 0, 140726605135552}}}}} keng = 0x0 evname = {s = 0x7ca474b9715a "timer", len = 5} __func__ = "timer_handler" --Type <RET> for more, q to quit, c to continue without paging-- #8 0x00005745eb49e3c4 in slow_timer_main () at core/timer.c:1105 n = 12 ret = 4294967295 tl = 0x7ca47b2f7950 i = 1010 __func__ = "slow_timer_main" #9 0x00005745eb3a85bd in main_loop () at main.c:1765 i = 16 pid = 0 si = 0x0 si_desc = "udp receiver child=15 sock=10.16.0.74:5101 ( 35.202.133.184:5101)\000AOw\375\177\000\000d\366\023\376\244|\000\000 \000\000\000\060\000\000\000@AOw\375\177\000\000\200@Ow \375\177\000\000\000B\213&T\004\207d\223\311\177\353EW\000\000\240\274\065\366\244|\000" nrprocs = 16 woneinit = 1 __func__ = "main_loop" #10 0x00005745eb3b34c3 in main (argc=8, argv=0x7ffd774f4688) at main.c:2942 cfg_stream = 0x5745ed74d2a0 c = -1 r = 0 tmp = 0x7ffd774f5d9f "" tmp_len = 0 port = 0 proto = 0 ahost = 0x0 aport = 0 options = 0x5745eb7b8d78 ":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 = 1555962608 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 2 n_lst = 0x7ffd77580248 p = 0x7ca4fe189995 <handle_intel+197> "H\205\300uf\213t$\004M\215\216\244" st = {st_dev = 0, st_ino = 0, st_nlink = 0, st_mode = 0, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 0, st_blksize = 0, st_blocks = 0, st_atim = {tv_sec = 0, tv_nsec = 0}, st_mtim = {tv_sec = 0, tv_nsec = 0}, st_ctim = {tv_sec = 0, tv_nsec = 0}, __glibc_reserved = {0, 0, 0}} tbuf = "\b\000\000\000\000\000\000\000\210FOw\375\177\000\000\320FOw\375\177\000\000\330+,\376\244|\000\000\000\000\000\000\000\000\000\000\020\vH\376\244|\000\000\001", '\000' <repeats 19 times>, " \000\000\000(&G\376\244|\000\000\320EOw\375\177\000\000\312tH\376\244|\000\000\000\000\000\000\000\000\000\000@e*\376\244|\000\000\320\263+\376\244|\000\000\364EOw\375\177\000\000\006", '\000' <repeats 15 times>, "/", '\000' <repeats 39 times>, "\200\037\000\000\377\377", '\000' <repeats 186 times>... option_index = 0 long_options = {{name = 0x5745eb7baf46 "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x5745eb7b6524 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x5745eb7baf4b "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x5745eb7baf51 "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x5745eb7baf57 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x5745eb7baf60 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x5745eb7baf6a "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x5745eb7baf74 "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x5745eb7baf7f "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x5745eb7baf88 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x5745eb7baf93 "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x5745eb7baf99 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main" (gdb) list 411 } 412 } while(RES_ROW_N(res) > 0); 413 414 for (i = 0; i < rd->carrier_num; ++i) { 415 for (j = 0; j < rd->carriers[i]->domain_num; ++j) { 416 if (rd->carriers[i]->domains[j]->sum_prob == 0.0) { 417 LM_ERR("All routes with carrier id %d (%.*s) and domain id %d (%.*s) have probability 0.\n", 418 rd->carriers[i]->id, rd->carriers[i]->name->len, rd->carriers[i]->name->s, 419 rd->carriers[i]->domains[j]->id, rd->carriers[i]->domains[j]->name->len, rd->carriers[i]->domains[j]->name->s); 420 } Please help to understand the reason for the crash and how to fix the problem. Thank you in advance. Regards *Maharaja Azhagiah*

3 7

Limiting Video calls(Volte to video conversion)
by dushy 28 Aug '24

28 Aug '24

Hello All, I am trying to limiting the number of video calls (converting Volte to Vilte). To reduce traffic on the enodeb. Its like if there are 10 volte calls going on then I should be able to limit the number calls that can be converted to Vilte. I tried to write a route in kamailio_pcscf.cfg using module ims_dialog. But it doesnt work and get some errors while doing a modparam. Tried another method of counting eRAB at eNB (ENBUEID and ERABID pair with qci-2). This worked, but after sometime the extra call got established. Looks like traffic is going through qci 9 bearer. Is there any method where we can achieve this. Thanks in Advance. Regards, Dushyanth

1 0

Re: certificate verify failed (sni: unknown) integration with ms teams
by Muhammad Sohaib 27 Aug '24

27 Aug '24

Hi Batanai, After disabling #!define WITH_NAT in Kamailio, calls from the Asterisk endpoint to Microsoft Teams are functioning correctly. Both inbound and outbound calls exhibit proper two-way audio and no disconnection issues but there is no Bye when hangup from ms teams. However, I am encountering a problem with outbound calls originating from Microsoft Teams. When a call is placed from MS Teams, the recipient's phone rings as expected. Once the call is answered, Microsoft Teams disconnects the call immediately. The root of the issue seems to be that the ACK message is not being sent or received properly. As a result, Microsoft Teams does not receive the ACK, leading to an immediate call termination. Additionally, Asterisk does not receive a BYE message from Kamailio. *Summary of the Issue:* - *Inbound Calls*: From Asterisk to MS Teams work fine, with full audio and no disconnections but no BYE from ms teams. - *Outbound Calls*: From MS Teams to Asterisk result in the call being disconnected immediately after being answered due to a missing ACK, and Asterisk does not receive a BYE message. Thanks, On Tue, Aug 27, 2024 at 5:45 PM Batanai Musiiwa <batanai.musiiwa(a)etel.co.zw> wrote: > Hi @ > Muhammad. May you advise if the calls are disconnecting in either > direction? > > ; > > On Tue, Aug 27, 2024 at 11:05 AM Muhammad Sohaib via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi everyone, >> >> Thank you all for your kind help with the TLS configuration. Palany's >> suggestion worked flawlessly, and now both inbound and outbound calls are >> functioning. >> >> However, I'm facing a one-way audio issue, and calls are disconnected >> after 30 seconds. I suspect this might be related to NAT. Here’s the >> current setup: >> >> *Asterisk <===> Kamailio (as SBC) <=====> MS Teams* >> >> Could this be related to my NAT configuration, or is there something else >> I should investigate? Any guidance would be greatly appreciated. >> >> Thanks again for your support! >> >> On Thu, Aug 15, 2024 at 11:02 PM palany <palany(a)advancedzim.com> wrote: >> >>> >>> >>> Hi Muhammad >>> >>> >>> >>> Can you try your tls cfg as below and make sure your certificates have >>> the right permissions. >>> >>> >>> >>> [server:default] >>> >>> method = TLSv1.2+ >>> >>> verify_certificate = no >>> >>> require_certificate = no >>> >>> private_key = /etc/letsencrypt/live/MYDOMAIN/privkey.pem >>> >>> certificate = /etc/letsencypt/live/MYDOMAIN/fullchain.pem >>> >>> >>> >>> [client:default] >>> >>> method = TLSv1.2+ >>> >>> verify_certificate = no >>> >>> require_certificate = no >>> >>> private_key = /etc/letsencrypt/live/MYDOMAIN/privkey.pem >>> >>> certificate = /etc/letsencrypt/live/MYDOMAIN/fullchain.pem >>> >>> *From:* Muhammad Sohaib via sr-users [mailto:sr-users@lists.kamailio.org] >>> >>> *Sent:* Thursday, 15 August 2024 3:46 PM >>> *To:* Kamailio (SER) - Users Mailing List >>> *Cc:* Muhammad Sohaib >>> *Subject:* [SR-Users] certificate verify failed (sni: unknown) >>> integration with ms teams >>> >>> >>> >>> Dear all, >>> >>> >>> >>> Trying to integrate Kamailio with MS Teams by following >>> https://skalatan.de/en/blog/kamailio-sbc-teams >>> >>> >>> >>> kamcmd dispatcher.list | egrep "URI|FLAGS" >>> >>> >>> URI: sip: >>> sip.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> URI: sip: >>> sip2.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> URI: sip: >>> sip3.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> >>> >>> >>> Kamailio Logs: >>> >>> /usr/local/sbin/kamailio[412158]: INFO: <script>: Sent out tm request: >>> OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012 >>> Via: SIP/2.0/TLS >>> x.x.x.x:5061;branch=z9hG4bK5dad.92de50b2000000000000000000000000.0#015#012 >>> To: <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012 >>> From: <sip:test.mytest.com >>> >;tag=5d0939b82abe9b1bbf185d963b6e6c88-edeb3c71#015#012 >>> CSeq: 10 OPTIONS#015#012 >>> Call-ID: 2db6bede5631d9b6-412158(a)88.99.244.106#015#012 >>> <http://2db6bede5631d9b6-412158@88.99.244.106#015%23012> >>> Max-Forwards: 70#015#012Content-Length: 0#015#012 >>> User-Agent: kamailio (5.8.2 (x86_64/linux)) >>> >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1312]: >>> tls_h_read_f(): protocol level error >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_util.h:49]: >>> tls_err_ret(): TLS write:error:0A000086:SSL routines::certificate verify >>> failed (sni: unknown) >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1316]: >>> tls_h_read_f(): src addr: 52.114.75.24:5061 >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1319]: >>> tls_h_read_f(): dst addr: x.x.x.x:0 >>> /usr/local/sbin/kamailio[412165]: ERROR: <core> [core/tcp_read.c:1524]: >>> tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fa74d265d40 r: >>> 0x7fa74d265e68 (-1) >>> >>> >>> tls.cfg: >>> >>> [server:default] >>> method = TLSv1.2+ >>> verify_certificate = yes >>> require_certificate = yes >>> private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem >>> certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> server_name = test.mytest.com >>> >>> [client:default] >>> method = TLSv1.2+ >>> verify_certificate = yes >>> require_certificate = yes >>> private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem >>> certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> >>> Please suggest what I am missing. >>> >>> --- >>> >>> Thanks, >>> >> >> >> -- >> Thanks, >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > -- Thanks,

1 0

certificate verify failed (sni: unknown) integration with ms teams
by Muhammad Sohaib 27 Aug '24

27 Aug '24

Dear all, Trying to integrate Kamailio with MS Teams by following https://skalatan.de/en/blog/kamailio-sbc-teams kamcmd dispatcher.list | egrep "URI|FLAGS" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP Kamailio Logs: /usr/local/sbin/kamailio[412158]: INFO: <script>: Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012 Via: SIP/2.0/TLS x.x.x.x:5061;branch=z9hG4bK5dad.92de50b2000000000000000000000000.0#015#012 To: <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012 From: <sip:test.mytest.com >;tag=5d0939b82abe9b1bbf185d963b6e6c88-edeb3c71#015#012 CSeq: 10 OPTIONS#015#012 Call-ID: 2db6bede5631d9b6-412158(a)88.99.244.106#015#012 Max-Forwards: 70#015#012Content-Length: 0#015#012 User-Agent: kamailio (5.8.2 (x86_64/linux)) /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1312]: tls_h_read_f(): protocol level error /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_util.h:49]: tls_err_ret(): TLS write:error:0A000086:SSL routines::certificate verify failed (sni: unknown) /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1316]: tls_h_read_f(): src addr: 52.114.75.24:5061 /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1319]: tls_h_read_f(): dst addr: x.x.x.x:0 /usr/local/sbin/kamailio[412165]: ERROR: <core> [core/tcp_read.c:1524]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fa74d265d40 r: 0x7fa74d265e68 (-1) tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem server_name = test.mytest.com [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem Please suggest what I am missing. --- Thanks,

5 5

I have mo media and I must park
by Alexandru Covalschi 27 Aug '24

27 Aug '24

Hello list! Please help me understand if putting my b2bua in sensory deprivation is actually the best approach to punish it for sole reason of its existence. The background of this horror story is pretty cliche, call center needs a way to infinitely expand, so I'm building a scaling telephony core. Our main protagonist is an innocent b2bua with a very grounded backstory: step 0: receive external instruction to start a call step 1: originate step 2: park step 3: play moh until: - - a) told to pass it to other component - - b) told to hang up - - c) hang up on local timeout A simplified diagram of the call flow looks something like this (repeating names are same components): ### Signaling ### Remote Destination ^ Kamailio-external ^ Freeswitch-originator V Freeswitch- queue V Kamailio-local V Agent ### end of signaling ### ### Media ### Remote Destination ^ Rtpengine-external ^ Freeswitch-originator V Freeswitch-queue V Rtpengine- local V Agent ### end of media ### Due to specifics of the setting, re-negotiating the media and any non-essential signaling with remote destination is a very much last resort. And, as you can imagine, after step3a (pass it to other component), the character development of Freeswitch-originator service stops completely. Well, at least it is doing an honest job by playing the music on hold, which a little bit justifies its presence on the media path. But not for long. In the world of near future, pull request number 3956 is merged with the master branch, exposing the rtpengine API and leaving me no reason not to move MoH to a kernel media player. At this moment I feel a wild urge to get rid of Freeswitch-originator at all, but then I read the sources for mohq and dialog to remind myself that Kamailio is a proxy and proxies can't do this... So I fill b2bua's dialplan with 387.44 million lines of hateful comments and remind it to rcvonly every time it tries to warn the others about the grim fate. When the queue designates an agent to his call, theoretically I don't even need the second freswitch as well - if agen't media session has been established (sometimes ICE take tooo long) with Rtpengine- local, I can bridge their media sessions using publish and subscribe. I'd like to ask your opinion, list: do you see any plotholes potential issues with such deisgn? I'd honestly like to set Freeswitch to bypass-media, so it wont' waste ports and cpu for nothing at all. Are there any issues if I add body for the iniitial invite in the request_route as if it was received in there? Maybe there's a better way, overall? Any critique is welcome! Thank you -- AC

1 0

How to invoke RTPengine for re-Invite 200 OK within dialog
by Shah Hussain Khattak 27 Aug '24

27 Aug '24

Hello Experts, I need your advice on the following issue if you can kindly help. I am working on a hold call flow: UA - > Kamailio (RTP Engine) - > FS initial call establishes fine, RTP engine is invoked and SDP media IP is modified correctly. Then 1. I am receiving a hold re-invite with a send-only 2. I see Kamailio is changing the media IP in the SDP (with my RTP engine IP) when it sends the Invite to my core FS server. 3. FS sends 200 OK with a=recvonly 4. Kamailio relays the message as it is to the other leg. can someone advise me, on how can I invoke rtpengine at step 4, so the SDP is sent with my media server IP in the 200 OK? --- 200 OK from FS ---- SIP/2.0 200 OK Via: SIP/2.0/UDP 13.54.01.02:5060;branch=z9hG4bKd2e8.8a573af8d927f67c1db4550676f8be88.0;i=1 Via: SIP/2.0/TCP 192.168.10.117:51280;received=218.33.01.02;rport=51280;branch=z9hG4bKPj7d1d8968830d4ed3922eec7da7cdc17c;alias Record-Route: <sip:13.54.01.02:5060;r2=on;lr=on> Record-Route: <sip:13.54.01.02;transport=tcp;r2=on;lr=on> From: <sip:0403225908@trunk.xyz.corp>;tag=6a180ffa7f824972a2c59ba47f293acb To: <sip:+61285038000@trunk.xyz.corp>;tag=DZrpcpgS1BDKF Call-ID: 4154062196634c34a2dbcb6b612bf44c CSeq: 12263 INVITE Contact: <sip:+61285038000@54.88.54.88:5060;transport=udp> User-Agent: UserAgentX/v1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1724635755 1724635757 IN IP4 54.88.54.88 s=FreeSWITCH c=IN IP4 54.88.54.88 t=0 0 m=audio 21202 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly a=ptime:20 a=rtcp:21203 IN IP4 54.88.54.88 --- 200 OK relayed towards the UA ---- 2024/08/26 17:22:39.923639 172.16.8.126:5060 -> 218.33.78.33:51280 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.10.117:51280;received=218.33.01.02;rport=51280;branch=z9hG4bKPj7d1d8968830d4ed3922eec7da7cdc17c;alias Record-Route: <sip:13.54.01.02:5060;r2=on;lr=on> Record-Route: <sip:13.54.01.02;transport=tcp;r2=on;lr=on> From: <sip:0403225908@trunk.xyz.corp>;tag=6a180ffa7f824972a2c59ba47f293acb To: <sip:+61285038000@trunk.xyz.corp>;tag=DZrpcpgS1BDKF Call-ID: 4154062196634c34a2dbcb6b612bf44c CSeq: 12263 INVITE Contact: <sip:+61285038000@54.88.54.88:5060;transport=udp> User-Agent: UserAgentX/v1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1724635755 1724635757 IN IP4 54.88.54.88 s=FreeSWITCH c=IN IP4 54.88.54.88 t=0 0 m=audio 21202 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly a=ptime:20 a=rtcp:21203 IN IP4 54.88.54.88 my configs for within dialog route: ##### route[WITHINDLG] { append_hf("X-Test-Header: route_WITHINDLG\r\n"); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if (is_method("ACK")) { xlog("L_INFO", "Relaying ACK\n"); } if (is_method("INVITE|UPDATE")) { if (has_body("application/sdp") && search_body("m=image") && search_body("T38")) { xlog("L_INFO", "T.38 re-INVITE detected, skipping rtpengine invocation\n"); } else { xlog("L_INFO", "Non-T.38 INVITE detected, invoking rtpengine\n"); rtpengine_manage("replace-origin replace-session-connection"); } } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); } exit; } } ##### looking forward to your usual guidance. Thank you! Regards, Shah

2 2

socket names for command line parameter "listen"
by Xenofon Karamanos 23 Aug '24

23 Aug '24

Hello all, I wanted to inform you about an upcoming change regarding the syntax we use for defining socket names in the listen (-l) command line argument. Current Syntax: -l udp:10.10.10.10:5060/11.11.11.11:5060 Proposed Options: 1. -l udp:10.10.10.10:5060/11.11.11.11:5060=s1 2. -l s1=udp:10.10.10.10:5060/11.11.11.11:5060 ( A draft PR about this option can be found in https://github.com/kamailio/kamailio/pull/3954) [https://opengraph.githubassets.com/69f8a4a1d2c8988372b4942f84bde95de94502d2…]<https://github.com/kamailio/kamailio/pull/3954> main: Add support for naming socket in cli by xkaraman · Pull Request #3954 · kamailio/kamailio<https://github.com/kamailio/kamailio/pull/3954> Pre-Submission Checklist Commit message has the format required by CONTRIBUTING guide Commits are split per component (core, individual modules, libs, utils, ...) Each component has a single... github.com Please note that the socket name (s1) in both options is/should be optional. The change will allow for better clarity and flexibility when defining sockets, particularly when a name needs to be assigned and match the config file style. I would appreciate your feedback on which syntax you find more intuitive or any concerns you might have with these options. Your input is important to ensure a smooth transition. Best regards, Xenofon

2 2

Kamailio cluster options
by Ali Taher 22 Aug '24

22 Aug '24

Hello all, I'm using Kamailio as SIP redirect (act as routing decision for SBC) depending mainly on sqlops and db_postgres modules, and it's working perfectly. What I need now is to have a kind of cluster of Kamailio servers for high availability purpose. What is the best option for me ? BR

2 1

push server tcp problem
by Bernd Krueger-Knauber 20 Aug '24

20 Aug '24

Hi, I implemented an apple and android push server/proxy with kamailio 5.8.1. The server/proxy is transparent. It works fine with UDP, since the registration after the push comes from the same ip and port. So t_continue works without problems. (We use the new registration as signal thet the phone is ready) If we use tcp at the client, he gets the push, wakes up and send the new registration on an other port. The SIP stack uses now an other connection. t_continue fails, because it uses the old data of the original invite acccording to the first registration. Is it possible to change the stored connection values according to the new registration data? Best regards, Bernd

2 2

low performance with udp_children and async_workers
by elhar.mohamed＠gmail.com 19 Aug '24

19 Aug '24

Hello, Based on business requirements, 35% of invites should be handled by async_workers and 65% with be handled by main children. The target is 1500cps. I'm using the following conf : children=64 async_worker=128 when doing performance tests using SIPp, I figured out the 1500cps is well supported by the conf above, but when I updated the data in database to not use anymore async workers, the cps is getting decreased and it was between 200 and 500. This a client requirements is to be able to enable/disable feature that needs to be handled by async workers. Could you please help me on this? Thanks in advance

4 12

Inbound call to webRTC failing with TCP/TLS connection (id: 0) for WebSocket could not be found
by hollis.rhonda＠gmail.com 16 Aug '24

16 Aug '24

We're seeing an issue where some webRTC clients are not receiving inbound calls. Kamailio logs show the following error: WARNING: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: <core> [core/msg_translator.c:3007]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: <core> [core/msg_translator.c:2086]: build_req_buf_from_sip_req(): could not create Via header ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_fwd.c:484]: prepare_new_uac(): could not build request ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_fwd.c:1764]: t_forward_nonack(): failure to add branches DEBUG: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_funcs.c:358]: t_relay_to(): t_forward_nonack returned error -2 (-2) DEBUG: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_funcs.c:376]: t_relay_to(): -2 error reply generation delayed It seems to be happening after the client has been connected to Kamailio for more than 24 hours. The socket connection details look correct in the external DB and in the output of ws.dump, core.tcp_list, and ul.dump. A restart of Kamailio or a reload of the webRTC client, triggering a new websocket connection, will clear the issue. However, we haven't been able to determine exactly when and why a client ends up in this state. While a client is in this state, REGISTER requests are still handled successfully, they can make outbound calls and TCP keepalives from the client are working. I'm still working on trying to debug this with GDB so I can see what the value of send_info is in https://github.com/kamailio/kamailio/blob/master/src/core/msg_translator.c#…, but I wasn't having luck with multiple child processes running, and now my client has reloaded and is working again. I've restarted Kamailio with a single child process, so when it does start failing again, I may have better luck with GDB. Any thoughts on what might be causing clients to get into this state? Whatever additional information I can provide that might help, I'm happy to share. Thanks!

2 4

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 16 Aug '24

16 Aug '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

3 2

[http_client] KEMI use "curl_connect_post" send http request error
by 17603019528＠163.com 16 Aug '24

16 Aug '24

ALL: Hi, here is my lua script: ``` local body = '{"ip":"172.16.4.111", "ports":[5261, 5260]}' local resp = "" local code = KSR.http_client.curl_connect_post("configapi", "", "application/json", body, resp) ``` the error is : ``` 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[0] for: curl_connect_post is str: configapi 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[1] for: curl_connect_post is str: 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[2] for: curl_connect_post is str: application/json 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[3] for: curl_connect_post is str: {"ip":"172.16.4.111", "ports":[5261, 5260]} 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[4] for: curl_connect_post is str: 24(2406) ERROR: <core> [core/pvapi.c:429]: pv_cache_get(): invalid parameters 24(2406) ERROR: http_client [http_client.c:739]: ki_curl_connect_post(): failed to get pv spec for: ``` I know the error means: "resp" type is not right, what is the correct type ? I search google for this issue, no one knows, someone please help me. Thinks

2 2

TLS module doesn't support TLSV1
by o.atef＠fiberme.com 15 Aug '24

15 Aug '24

Hello Brothers, I've installed kamailio throw "apt install" on Debian and it's installed version: kamailio 5.6.3 (x86_64/linux). Now I've a big problem that kamailio cannot running with TLSv1 and it has to be TLSv1.2+ as tls.cfg doc said: # We do not enable anything else than TLSv1.2+ # over the public internet. Clients do not have # to present client certificates by default. How could I avoid this restriction please to enable TLSv1? Thank you,

5 5

[PHONENUM] Match with country on Lua/KSR
by João Silva 14 Aug '24

14 Aug '24

Hello everyone! The function phonenum_match_cn from module PHONENUM is not available on KSR. Is there any work around to use it on Lua config? Best regards, Joao

2 2

Incoming call with one one way audio and ctting off after thirty seconds
by palany 14 Aug '24

14 Aug '24

I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as my SBC. Calls from freepbx to teams are working fine with audio and I am having challenges with calls from teams to freepbx. The calls rings and after answering there is only way audio and the call drops off after thirty seconds. My pbx is behind a microtik router and I have port forwarded my sip port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is hosted in AWS. I have attached both my pcap from Kamailio, freepbx and my Kamailio.cfg. I am thinking It's a misconfiguration challenge or something else. Please help I am failing to see my challenge. Best Regards,

3 3

Authentication
by Vishal Pai 13 Aug '24

13 Aug '24

Hello All I am using kamailio sip account as gateway at Freeswitch side but when I enable caller id from = true then call from Freeswitch to kamailio didn’t authenticate As I check it’s about from header when I disable caller id from = false the call get authenticate since it get sip username and domain but when there is caller id the from header get callerid@ip address Can anyone guide me Thanks

2 1

Kamailio hangs on get_profile_size method.
by satyaprakash ch 12 Aug '24

12 Aug '24

HI, We are facing a hang issue with Kamailio for the get_profile_size method. Randomly, the Kamailio got hangs, and the CPU went 100%. During the hanging time, we generated the TRAP and we could see that the hang happened on the get_profile_size method. We are getting this hang randomly in dlg_profile.c of the following lines: The hang occurs, causing the CPU to spike to 100% at one of the following lines randomly: 1. if(value->len == ph->value.len && memcmp(value->s, ph->value.s, value->len) == 0) { 2. ph = ph->next; 3. while(ph != profile->entries[i].first); During our analysis, we found that PROCESS ID 7608 gets a spike of 100% utilization. We suspect there may be an issue with the processing of the 'ph' variable, but we couldn't identify the root cause. I've attached the Kamailio TRAP for further details. Could you please suggest what might be causing this problem and how we can resolve it? Thank you.

1 1

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 10 Aug '24

10 Aug '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria [aws](https://www.igmguru.com/cloud-computing/aws-developer-certification-tr…

4 4

Kamailio hangs on get_profile_size method.
by satyaprakash ch 09 Aug '24

09 Aug '24

HI, We are facing a hang issue with Kamailio for the get_profile_size method. Ramdomly, the Kamailio got hangs, and the CPU went 100%. During the hanging time, we generated the TRAP and we could see that the hang happened on the get_profile_size method. We are getting this hang randomly in dlg_profile.c of the following lines: The hang occurs, causing the CPU to spike to 100% at one of the following lines randomly: 1. if(value->len == ph->value.len && memcmp(value->s, ph->value.s, value->len) == 0) { 2. ph = ph->next; 3. while(ph != profile->entries[i].first); During our analysis, we found that PROCESS ID 7608 gets a spike of 100% utilization. We suspect there may be an issue with the processing of the 'ph' variable, but we couldn't identify the root cause. I've attached the Kamailio TRAP for further details. Could you please suggest what might be causing this problem and how we can resolve it? Thank you.

1 0

Incoming call with one one way audio and cutting off after thirty seconds
by palany 08 Aug '24

08 Aug '24

I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as my SBC. Calls from freepbx to teams are working fine with audio and I am having challenges with calls from teams to freepbx. The calls rings and after answering there is only way audio and the call drops off after thirty seconds. My pbx is behind a microtik router and I have port forwarded my sip port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is hosted in AWS. Below are google drive links logs for the incoming call in Kamailio , freepbx and my Kamailio.cfg. I am thinking It's a misconfiguration challenge or something else. Please help I am failing to see my challenge. kamailio.pcap - https://drive.google.com/file/d/1aoJhksgwcH-52unwxuyzn8GAmG9PEnOM/view?usp=s haring freepbx.pcap - https://drive.google.com/file/d/1JJXs9UOUFak5Y4uWI_ARB56vQfVYfSDj/view?usp=s haring kamailio.cfg - https://drive.google.com/file/d/1v2ZtylOf8fIMYH4OckGmnP489PO7WI7J/view?usp=s haring Best Regards

2 2

Dialog timer reset on UPDATE
by Alex Balashov 08 Aug '24

08 Aug '24

Hi, I have a dialog where I have set: dlg_set_property("timeout-noreset"); But it seems the timeout is reset anyway when handling an in-dialog UPDATE request, at least, on Kamailio 5.5.4. This does not seem to happen with reinvites. I haven't been able to get more information thus far, other than to be able to say that there is nothing unusual about the UPDATE request and there are no additional errors at log level 2. I was wondering if anyone else has encountered this. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

1 0

app_python3 method not found or is not callable
by Westhuis, Johannes 08 Aug '24

08 Aug '24

Hi, we are using kamailio with app_python3. It works very good, but in some cases, there are error logs stating: app_python3 [python_exec.c:91]: apy_exec(): <python method> not found or is not callable It happens sporadically, and a restart of kamailio fixes it. As far as i can see it might be linked to concurrent arrival of several messages. Best regards, Johannes # kamailio -v version: kamailio 5.5.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: compiled on 12:01:03 Jul 22 2024 with gcc 10.2.1

1 0

http_async_query on carrierfailureroute changes rd back to carrieroute primary
by Maharaja Azhagiah 07 Aug '24

07 Aug '24

Hi I am running kamailio 5.4 on debian I have carrierfailureroute configured incase of primary service provider fails. I also have Stirshaken configured to add Identity header on outbound calls. Issue is when call fail overs to carrierfailureroute, http_async_query changes $ru to the primary carrier From the debug logs, when primary carrier sends a 488 (primary carrier expects SIP TLS but my call is UDP - to test the failover scenario) 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} tmx [t_var.c:561]: pv_get_tm_reply_code(): reply code is <488> 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:178]: set_next_domain_on_rule(): searching for matching routing rules39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:197]: set_next_domain_on_rule(): next_domain is 47987 Carrier route rewrites the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:706]: cr_do_route(): uri 14371234567 was rewritten to sip:14371234567@sip.primaryprovider.com, carrier 1, domain 47987 Before http_async_query rd and ru are still the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2976] - Debug testing ----- rd is sip.primaryprovider.com ----- ru is sip:14371234567@sip.primaryprovider.com 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:469]: async_send_query(): transaction suspended [5261:1830449764] 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:625]: async_push_query(): query sent [ https://authn-uat.ccid.neustar.biz/ccid/authn/v2/identity?apiKey=randomkey] (0x7fdcad097e60) to worker 1 However, when the route is being called after the http_async_query it changes to the primary one: 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) DEBUG: http_async_client [async_http.c:235]: async_http_cb(): resuming transaction (5261:1830449764) 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) INFO: <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2995] - Debug testing ----- rd is 1.2.3.4 ----- ru is sip:14371234567@1.2.3.4:5061;transport=TLS Due to this, call keeps going to the primary and it fails if ( http_async_query(STIRSHAKEN_AS_URL, "AS_RESPONSE") == -1 ) { xlog("L_ERR ", "[cfg:$cfg(line)] Failed to connect AS service for token $fu -> $tu \n"); return; } route[AS_RESPONSE] { xlog("L_INFO", "[callid: $ci] - [cfg:$cfg(line)] - Debug testing ----- rd is $rd ----- ru is $ru\n"); if ($http_ok) { xlog("L_INFO", "[cfg:$cfg(line)] Resuming outbound call transaction for $fu -> $tu Received - $http_rb \n"); # Add identity and Date headers if (jansson_get("identity", $http_rb, "$var(identity)")) { insert_hf("Identity: $var(identity)\n", "Content-Length"); } if (jansson_get("date", $http_rb, "$var(date)")) { if ($hdr(Date) != $null){ remove_hf("Date"); } insert_hf("Date: $var(date)\n", "Identity"); } } else { xlog("L_ERR", "[cfg:$cfg(line)] Resuming outbound call transaction. Error - $http_err)\n"); } route(RELAY); exit; } Please help to understand why rd / ru changes to primary carrier. Regards, Maharaja Azhagiah

3 7

Kamailio hang Issue
by satyaprakash ch 06 Aug '24

06 Aug '24

Hi, We are encountering a hang issue after deploying the following code: *route(DLG_MANAGE);* *if ($dlg_var(voip_id) == $null) { $dlg_var(voip_id) = $hdr(X-Voip-ID);}route[DLG_MANAGE] { if (is_method("INVITE")) { dlg_manage(); }}* This code includes the set_dialog_profile. We have this setup with 2 proxy servers with DMQ enabled, but the issue only occurs on one proxy. When the Kamailio proxy hangs, no logs are written. If you have any suggestions to resolve this issue, please help. Thanks.

2 2

Kamailio-IMS: After a abornamal Call, other call is not working (even after a new registration)
by deep.upreti＠gmail.com 05 Aug '24

05 Aug '24

Hello, Can anyone help me to resolve one issue? scenario is - 1- Register two UE (UE-A and UE-B) 2- Call from UE-A to UE-B for a specific duration (lets say: 100 seconds) 3 Verify if call is working: Answer is YES, call is working. 4. shutdown the eNB 5. Observation: No BYE/De-registration message is not coming as expected. 6. Again start eNB and register same UE (UE-A and UE-B) 7. Register UE-A and UE-B both 8. It will not remove previous contacts or not updating the reg-state. 9. I am running with Rx enable mode.

1 1

Multiple TLS connections to the same IP:Port
by Nick Digalakis 05 Aug '24

05 Aug '24

Hello everyone, I am trying to use a single Kamailio server to register to multiple MS Teams Direct Routing endpoints. The config snippet I am using is this: sht_iterator_start("i1", "teams-endpoints"); while(sht_iterator_next("i1")) { $var(teams_endpoint) = $shtitkey(i1); $xavp(tls=>server_name) = $var(teams_endpoint); $xavp(tls[0]=>server_id) = $var(teams_endpoint); $uac_req(method)="OPTIONS"; $uac_req(ruri)="sip:sip.pstnhub.microsoft.com:5061;transport=tls"; $uac_req(furi)="sip:" + $var(teams_endpoint); $uac_req(turi)="sip:sip.pstnhub.microsoft.com:5061;transport=tls"; $uac_req(hdrs)="Contact: <sip:" + $var(teams_endpoint) + ":" + "5061" + ";transport=tls>\r\n"; ### Create a unique Call-ID based on the Timestamp and the Message Body in MD5 $var(unhashed_cid) = $TV(Sn) + $mb + ""; $uac_req(callid)=$(var(unhashed_cid){s.md5}); uac_req_send(); } sht_iterator_end("i1"); When the HTable has only one endpoint, everything works fine. When I add a second endpoint, the first one continues to work but the second one failing with the error from Microsoft: Q.850;cause=63;text="85babcde-e0b5-4a85-8f4a-12345678c9ae;SBC certificate is not issued correctly. Provided trunk FQDN 'endpoint-02.domain.com' is not included in certificate's CN or SAN list. Certificate allows following FQDNs only: endpoint-01.domain.com. After some digging around, I realized that all endpoint after the first fail because Kamailio is re-using the same TLS connection for all subsequent OPTIONS as well, but of course the connection has been established with the certificate of the first endpoint. I have tested the certificates by switching around the first endpoint, so that shouldn't be a problem. Is there any way I can force it to establish a new TCP/TLS connection for each subsequent request? Any help would be much appreciated, I have been pulling my hair out with this one! Best regards, Nick [https://s-install.avcdn.net/ipm/preview/icons/icon-envelope-tick-green-avg-…]<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free.www.avg.com<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…>

2 2

How to log datetime in files
by 陈子妍 04 Aug '24

04 Aug '24

Hello! I am working on with Kamailio, I have already understood to add log prefix like `log_prefix="NAT {$rm $mt $hdr(CSeq) $ci}: "`, but I also want to record event time, so how can I config `datetime` in log files. Thanks~

3 2

Why does $T(id_index) return 0 even though t_suspend() is executed successfully?
by mm e 01 Aug '24

01 Aug '24

if(t_suspend()) { $dlg_var(start_time) = $timef(%Y/%m/%d %H:%M:%S); xlog("L_NOTICE", "==> processing suspended in transaction [$T(id_index):$T(id_label)] "); } else { xlog("L_ERR", "t_suspend err"); exit; } Sometimes, I get this log==> processing suspended in transaction [0:13614] "，as far as I know, 0 means that the value cannot be obtained. Executing t_continue(tindex, tlabel, rtname) with parameter 0 will fail.by the way, It's a question of probability.

1 0

Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 31 Jul '24

31 Jul '24

Hello Daniel, Regarding your question, I think only the avp(msg) should be destroyed as others avp's can be used elsewhere in the script. I tried the lastest git master and when restarting kamailio I got a segmentation fault: Jul 18 21:17:16 labcom070 systemd[1]: Started Kamailio - the Open Source SIP Server. Jul 18 21:17:17 labcom070 kernel: kamailio[2285]: segfault at 0 ip 00000000007eb66a sp 00007ffd7f80cc00 error 6 in kamailio[400000+533000] Jul 18 21:17:17 labcom070 kernel: Code: 41 56 41 55 41 54 53 48 81 ec c8 00 00 00 48 89 bd 48 ff ff ff 48 89 b5 40 ff ff ff 48 89 95 38 ff ff ff 48 8b 85 48 ff ff ff <48> c7 00 00 00 00 00 48 8b 85 48 ff ff ff c7 40 08 00 00 00 00 48 Jul 18 21:17:17 labcom 070 systemd[1]: Started Process Core Dump (PID 2286/UID 0). Jul 18 21:17:18 labcom070 systemd-coredump[2287]: Resource limits disable core dumping for process 2285 (kamailio). Jul 18 21:17:18 labcom070 systemd-coredump[2287]: Process 2285 (kamailio) of user 993 dumped core. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Main process exited, code=dumped, status=11/SEGV Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Failed with result 'core-dump'. Jul 18 21:17:18 labcom070 systemd[1]: systemd-coredump(a)6-2286-0.service: Succeeded. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 1. Jul 18 21:17:18 labcom070 systemd[1]: Stopped Kamailio - the Open Source SIP Server. Jul 18 21:17:18 labcom070 kernel: kamailio[2298]: segfault at 0 ip 00000000007eb66a sp 00007ffcd101c240 error 6 in kamailio[400000+533000] Jul 18 21:17:18 labcom070 kernel: Code: 41 56 41 55 41 54 53 48 81 ec c8 00 00 00 48 89 bd 48 ff ff ff 48 89 b5 40 ff ff ff 48 89 95 38 ff ff ff 48 8b 85 48 ff ff ff <48> c7 00 00 00 00 00 48 8b 85 48 ff ff ff c7 40 08 00 00 00 00 48 Regards, Vanderlei ________________________________ De: Daniel-Constantin Mierla <miconda(a)gmail.com> Enviado: quinta-feira, 18 de julho de 2024 02:51 Para: Vanderlei Torres Batistela <v.batistela(a)hotmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Assunto: Re: [SR-Users] Possible Kamailio memmory leak - version 5.8.2 Hello, indeed the previous commit was destroying a single avp, I pushed a new one to destroy all of them with that name. I am wondering if all avps, no matter the name should be destroyed. Anyhow, could you fetch the latest git master version and try your scenario again? Cheers, Daniel

1 0

[TM] [ASYNC] : Freeing transaction after sending 302 response
by elhar.mohamed＠gmail.com 31 Jul '24

31 Jul '24

Hello, I'm using the async_task_group_route for async module to handle some invites, when testing with sipp tools, I got some Dead call error message and after investigating It seems that for the same invite I sent multiple 302 message to sipp because the ACK sent by sipp was not handle before thr retr_timer1 & retr_timer2. Please can you help me to free the transaction after sending the first 302 ? Thank you in advance.

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

1 0

Job: System Engineer VoIP - Germany / HO
by Gerrit Jacobsen 25 Jul '24

25 Jul '24

VIER is looking for a System Engineer VoIP What VIER does VIER is a global contact center technology provider. VIER develops and runs its own technology in several data centers. You will be part of the Real-time Media Processing Team which is responsible for the telephony connections between carriers, applications and call-center agents. Your mission You are responsible for the support of the VoIP systems as well as the associated infrastructure Since you have sovereignty over the VoIP systems, you are also responsible for performing system upgrades and configurations. You have a lot of freedom to design, plan, and implement new VoIP system components for internal and external projects. You manage and implement change requests and are a specialist for special topics in the communication area. You show a high-sense of responsibility and foresight to keep systems running 24/7 You are prepared to work occasionally also in the evening hours to roll out new features Your skills You have several years experience in the day-to-day operation of telecom systems You have worked extensively with opensource VoIP software such as Kamailio / OpenSIPS, FreeSWITCH You can administer Linux systems such as RockyLinux and Debian You know SIP, RTP, RTCP, TLS, WebRTC protocols and can debug problems with Wireshark You have also worked with MySQL, Git You are communicative and like to work with team colleagues Further beneficial skills Experience with Ansible or Puppet for automation and configuration management. CheckMK, Homer, Hepic, Redis, CGRates, CDR Tools, Python Own VoIP software projects German language skills Please apply here: https://vier.jobs.personio.de/job/1638419?language=en&display=en

1 0

RTJSON: No Json for routing
by felipe.nunes＠comuniki.me 24 Jul '24

24 Jul '24

Good afternoon I have the following error in the application: [rtjson_routing.c:701]: rtjson_update_branch(): no json for routing This error occurs regardless of whether there is high demand or not and the application sending the data correctly. Can you help me with this? Felipe Nunes

2 1

ds_select from xhttp:request
by Christian Berger 24 Jul '24

24 Jul '24

Hi, we want to execute ds_select from an xhttp:request event route. The idea is to be able to get all active targets of a dispatcher set so our monitoring can check that via http which would be convenient for us. So are there any reasons why ds_select wouldn't work in an xhttp:request even route? Regards Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

3 2

Listening on multiple TLS ports
by João Silva 23 Jul '24

23 Jul '24

Hello everyone! Is there a way to define a listen port range for TCP/TLS? Or any macro I can use to create the LISTEN commands? And finally is there a performance penalty for having multiple (2k) listening ports at the same time? I am in a strange situation where I need to register multiple credentials on a single SIP trunk, for each registration I need to use a different local port otherwise the trunk will simply overwrite the previous registrations. Furthermore, when I place a call on that trunk I have to be consistent with the port I used for registration for that specific credential. So I am looking into explicitly defining the port to be used per credential, UACREG allows me to use the contact_addr/socket for the registration part and for placing of outbound calls I can use $fsn to force the correct socket. Best regards, Joao

3 4

IMS PCSCF force source IP address
by Anthony Blandin 23 Jul '24

23 Jul '24

Hello, I am using Kamailio pcscf with ipsec. I have one ipsec listen address which is used by the UE to connect to the PCSCF and one other listen address on port 5060. During the registration until the ipsec is not established, the UE sends the requests to the ipsec listen address and the pcscf is using the other listen address to forward the sip request to the ICSCF. That's is ok for me. But when the ipsec is established after the 200 OK REGISTER, the UE sends a SUBSCRIBE but the pcscf forwards the SUBSCRIBE to the SCSCF using its ipsec listen address instead of the other listen address. I tried to use $fs to force the socket but without success. Do you know how to force the source IP address to be used on the pcscf? Thanks for your support. Regards, Anthony

1 0

mysql error after migration from 5.7.5 to 5.7.6
by Alejandro Guralnik 23 Jul '24

23 Jul '24

Hello, After upgraded from kamailio 5.7.5 to 5.7.6, when did start kamailio process, got the following error: WARNING: db_mysql [km_my_con.c:179]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring After downgrade to 5.7.5 dind't happened. (we don't use mysql ssl connection).

2 3

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…

1 0

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria <a href="https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…">aws</a>

1 0

SHM Memory leak?
by Ihor Olkhovskyi 23 Jul '24

23 Jul '24

Hello! I'm trying to get where all the SHM memory gone. For the moment I'm using these settings: -m 8192 -M 1024 which means 8G of SHM meory. Kamailio parameters are the following fork=yes children=8 tcp_children=12 enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 And having ~1000 TLS/WSS clients I'm getting this after 3 days of running: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 2796", "shmem:free_size = 3737947072", (3.7 G) "shmem:max_used_size = 4857418512", (4.8 G) "shmem:real_used_size = 4851987520", "shmem:total_size = 8589934592", "shmem:used_size = 4838988096" (4.8 G) ], "id": 984479 } Means half of SHM memory is gone When running kamcmd mod.stats all shm I'm getting (output truncated for the largest values) Module: core { ... tcpconn_new(1201): 130669280 (0.13 G) ... Total: 131657632 (0.13 G) } Module: sl { ... Total: 7520 } Module: siptrace { ... Total: 13520 } Module: rr { Total: 0 } Module: dialog { ... Total: 146080 } Module: permissions { ... Total: 62144 } Module: htable { .... Total: 3359552 (3.3 M) } Module: rtpengine { ... Total: 31456 } Module: textopsx { Total: 0 } Module: tsilo { ... Total: 75072 } Module: tm { .... Total: 3459504 (3.4 M) } Module: usrloc { .... Total: 1217616 } Module: pua_dialoginfo { ... Total: 8720 } Module: pua { ... Total: 150848 } Module: pike { ... Total: 9504 } Module: websocket { wsconn_add(198): 54139856 (54 M) ... Total: 54148096 (54 M) } Module: debugger { ... Total: 21872 } Module: tmx { Total: 512 } Module: kex { .... Total: 1888 } Module: tls { ... ser_malloc(364): 87246112 (87 M) ... Total: 87997168 (87 M) } Module: secfilter { ... Total: 768 } Module: exec { ... Total: 16 } Module: dispatcher { ... Total: 2992 } Module: cfgutils { ... Total: 48 } Module: app_python3 { ... Total: 32 } So, I'm guessing, where all the memory gone? Is there any way to get more detailed info on SHM? Kamailio 5.8.2 -- Thanks in advance, Ihor

2 6

Not responding to OPTIONS if existing tcp connection
by Jon Bonilla (Manwe) 22 Jul '24

22 Jul '24

Hi all version 5.7.6 Debian 12 I'm having issues with a kamailio server not sending 200 to some OPTIONS requests. It does send 200 to some subscribers but not 100% of the time to 100% of subs. I have enabled debug=3 in that server and this is what I found: If the log says that it found existing tcp connection and it's reusing it I can't see the 200ok in a sngrep capture. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 2 <core> [core/tcp_main.c:2614]: tcpconn_send_put(): send from reader (360359 (18)), reusing fd <core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending... <core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c= 0x7fb0faa893d8 n=384 fd=14 <core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf=#012SIP/2.0 200 Keepalive#015#012Via: SIP/2.0/TCP 10.35.190.105:5060;branch=z9hG4bK15ebf764;rport=55894;received=1.1.1.186#015#012From: "Unknown" <sip:contesta01@10.35.190.105>;tag=as5311c3a7#015#012To: <sip: 1.1.1.138>;tag=054524a74bcccf444a0f08f67c5b7657.d07b5401#015#012Call-ID: 15dcd38c46f915576ded132a11d8ff4c@10.35.190.105:5060#015#012CSeq: 102 OPTIONS#015#012Server: aaa SBC#015#012Content-Length: 0#015#012#015#012 <core> [core/receive.c:531]: receive_msg(): request-route executed in: 302 usec On the other side, if kamailio doesn't find an existing tcp connection (at least nothing in the logs about that) I can see the 200 ok in sngrep. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/receive.c:531]: receive_msg(): request-route executed in: 295 usec All subs use tcp. And the kamailio.cfg is very simple regarding OPTIONS: Sanity, check maxfwd and sl_send_reply(200). Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

Drouting using TLS and domain names
by SAMUEL MOYA TINOCO 19 Jul '24

19 Jul '24

Hi everyone, I'm trying to configure drouting module but i'm not sure how to configure dr_gateways table to use domain names and TLS instead of IPs How should I do it? In dispatcher module is by using sips:domain but when I configure it like this I have problems to send invites (maybe I'm doing something wrong). Does anyone know how to configure it? Thanks in advance for your help Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DAD156.D8CFF170] Soluciones inteligentes para la autonomía personal

2 3

Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 18 Jul '24

18 Jul '24

Hello all, I am using a very simple scritp with corex module enabled . Normal calls works fine, but a scenario where INVITE does not receive response and the call is released with SIP 408 (request timeout) leads to memmory leak. I verified it making a lot of calls with this scenario and printing the shared memory (using kamcmd core.shmmem) . Its ocurred with modparam("corex", "nio_intercept", 1). If I modify to modparam("corex", "nio_intercept", 0) the memmory leak does not ocurr anymore. The problem also occurs in another scenario where the SIP Cancel message is not responded to. My script has this event route # event route event_route[network:msg] { xlog("L_INFO", "Dentro do Event_Route \n"); if (is_incoming()) { xlog("L_INFO", "Received message '$mb' \n"); $avp(msg) = $mb; xinfo("avp(msg) = $avp(msg)"); } else { xlog("L_INFO", "Sending message '$mb' \n"); $avp(msg) = $mb; }; } Test done wtih 4000 calls: [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 } I attached my kamailio script used and a log with one call using the command "kamcmd corex.shm_summary" with modparam("corex", "nio_intercept", 1) and modparam("corex", "nio_intercept", 0) Regards, Vanderlei

3 6

Re: Difference betwwen TM functions in KEMI
by Duarte Rocha 18 Jul '24

18 Jul '24

Hello Alex, You're totally right, that was the issue. Thank you so much for the help. Helpful as always. Cheers, Duarte Rocha

1 0

Difference betwwen TM functions in KEMI
by Duarte Rocha 17 Jul '24

17 Jul '24

Hello all, I'm using Kamailio with native script and KEMI in Javascript. I noticed that functions from the TM module can have different results if I call them on KEMI or Native script. Is this intended behaviour? Don't they share the state? In order to test this i made the following code : KEMI Code : function ksr_request_route() { try{ if ( KSR.is_method("BYE") ) { print.error("BYE Request KEMI BYE") KSR.route("exists_test") if(KSR.tm.t_exists()) { print.error("KEMI : Transaction exists") } else { print.error("KEMI : Transaction does not exist") } Native Code : route[exists_test] { if( t_exists()) { xerr("Native Script : Transaction exists \n"); } else { xerr("Native Script : Transaction does not exist \n"); } } I've also tried to change function call order but every time I get "Transaction does not exist" on native and "Transaction Exists" on KEMI. Is this intended behaviour? Should I only call tm functions in native or Kemi in order to have consistency ? Thanks in advance, Best Regards,

2 1

Dns record not alone?
by Pyry Aaltonen 17 Jul '24

17 Jul '24

Hi, I recently updated my testing env to 5.8.1 and after that I’ve been receiving dns_cache.c warning “record not alone”. I found out that it’s been added to quite recent commit: https://github.com/kamailio/kamailio/commit/d8a35b3b6c837b36779e232b65fce61… And that’s probably the why I haven’t seen it before. As far as I know, I have no other problem with dns or anything else than the warning appearing in the logs 😊 I can’t quite get what the warning is about? Clearly refers to dns queries, and with debug I found out that maybe its somehow related to ip which has several domains to pointing it (and queried by one of those domains by this kamailio). Can it be something with the PTR-record? And this warning comes ones every hour, so probably when some ttl expires, and meanwhile the cache is ok with the situation. And as a background, I haven’t done any specific dns-configurations in my config, so settings should be pretty much in defaults. Any advice what I should fix here to get the warning disappear? -Pyry

2 1

Add a Header to CANCEL request
by sadik.oualla.mohamed＠gmail.com 17 Jul '24

17 Jul '24

Hello all, I am facing an issue with Kamailio when trying to add custom headers to CANCEL requests. My current configuration adds the "Reason" header perfectly: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); } ``` However, when I attempt to add headers like "X-Reason" or "TestHeader", they are not added: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("X-Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); # or append_hf("TestHeader: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); #nothing changes in the relayed CANCEL request :-/ } ``` I've read in this discussion "https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…" that this might work if sending CANCEL in stateless mode, but I want to keep it stateful. Any suggestions on how to achieve this? Regards, Mohamed.

1 0

Changing ksr_tcp_accept_iplimit from default 1024
by Who AmI 16 Jul '24

16 Jul '24

Hi list, I hope you are all well? I am currently loadtesting some services internally and I am getting an odd error around TCP connections from the same source IP below: "/usr/sbin/kamailio[1806789]: CRITICAL: <core> [core/tcp_main.c:4459]: handle_new_connect(): hit the limit of connections per source IP (my_ip:39043) - rejecting" Its from the core the error so I was looking at the cookbooks and documentation but cannot get around this specific limit or find a configuration to tweak that fixes it. The limit only affects TCP (will affect TLS as well but I haven't got that far) and is 1024 it seems. Kamailio version is: 5.7.6 OS: Ubuntu 22.04 Things I have done so far: I have set limitNOFILE=65535 in the service file Set the tcp/tls_max_connections at 8192 (default is 2048) Increase the tcp_rd_buf_size to 16k+ The service I am using to loadtest is SIPp with flags "-t tn" for TCP multisocket as this service will have much more than 1024 concurrent TCP connections to it from a single source IP. I think I have missed something basic here as the devel and the source code references: "ksr_tcp_accept_iplimit" here at line 541: https://github.com/kamailio/kamailio/blob/master/src/main.c#L541 I cannot find a way to change this with a configuration line it seems, am I being blind or is my only option to build from source and change this? Many Thanks, John.

2 2

Weird telephone-event payload issue - need help
by Tim Bowyer 16 Jul '24

16 Jul '24

Hello Everyone, I've been grappling with this issue for some months, time to raise the white flag. Essentially I need to work out a way to configure Kamailio/RTPengine to normalise telephone-events to 101 earlier in the piece. 1. Call comes in from Carrier with INVITE w/SDP, telephone-event payload of 96. 2. INVITE sent to Asterisk with the same SDP, payload = 96 3. INVITE sent from Asterisk to Client with payload = 101 * 200 OK w/SDP from Client has payload = 101 * 200 OK w/SDP to Kamailio has payload = 96 * 200 OK w/SDP to Carrier has payload = 96 4. Asterisk (with Direct Media enabled) sends INVITE to Kamailio with payload = 96, and new media IP of Client 5. Kamailio passes this onto Carrier with media IP of RTPengine 6. 200 OK w/SDP from Carrier has payload = 96 sent to Kamailio 7. Kamailio passes this 200 OK to Asterisk, which responds with ACK and doesn't do anything further +---------+ +----------+ +---------+ +---------+ | Carrier | <----1-----> | Kamailio | <-----2----> | Asterisk| <----3------>| Client | | | | | | | | | +---------+ +----------+ +---------+ +---------+ ^ | | | | | |3| | | | | | +------5---------> +-----------+ <-----------------+------------4-----------+ | RTPengine | +-----------+ Big issue is, the Client is expecting telephone-events with a payload of 101, whereas RTPengine is passing it 96 from the Carrier. * Inbound DTMF only works at client-side if Asterisk stays in path (far from ideal). * The issue is exacerbated when calls are sent back out to the same or another carrier (forward scenario). * There's nothing I can seem to do in Asterisk to change how it handles DTMF in a direct media/re-invite setting. Any ideas? Cheers, Tim Bowyer

3 12

how to upgrade kamailio minor release from 5.7.5 to 5.7.6
by sadik.oualla.mohamed＠gmail.com 15 Jul '24

15 Jul '24

I am currently using Kamailio version 5.7.5, which I installed from the source code. I would like to upgrade to version 5.7.6 but have been unable to find documentation specifically addressing minor upgrades between these minor releases. I have found only wikis detailing the process for upgrading for major releases (like 5.7.x to 5.8.0). Could you please point me to the documentation for my current need or share with me steps on how to perform this minor upgrade? Thank you. Cheers.

3 4

How to get the rtpengine IP used by the current call among multiple rtpengines
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine

1 0

How to get the id or ip in the multiple rtproxies for LB with weights。
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine kamailio selected for the call.

1 0

Re: (no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi Brett! From what I could understand, my client (a Carrier) assigns services and products to its customers identified by trunkgroups (depending on the customer and the service requested, as well as the ingress trunkgroup the call will be forwarded to a specific egress trunkgroup). . The INVITE is sent by customer to the SBC (ingress), where some kind of process (which i have no details) will decide what IP:Port to forward the messages to Kamailio with additional infos (Kamailio is listening on different NICs/IPs/Ports). Then, Kamailio will check on the (external) routing engine the route to take and forwards to the outbound endpoint. Depending on the IP:Port that Kamailio receives the call (SBC may choose one NIC or the other) the request is then received by Kamailio and verified/processed by the routing engine replying back to kamailio with the route to take, and then Kamailio forwards to the outbound endpoint (egress SBC...). It is a bit confusing, and I want to simplify things as much as possible, but apparently the old/current solution is set this way and they pretend to keep the new solution as similar as possible to the old/current solution. This is mostly due to the billing process, which is very (overly?) complex and they do not pretend to change it. IMHO, the routing engine should be much simpler but it is a much more complex process the Carrier currently has...and wishes to keep. (sorry for posting without a subject. I did, however, re-posted with corrected subject) Atenciosamente / Kind Regards / Cordialement, *Sérgio Charrua* On Fri, Jul 12, 2024 at 8:11 PM Brett Nemeroff <brett(a)voicefoxtelephony.com> wrote: > Is there a reason you have to identify the trunk group based on received > IP/Port? I've seen this as a requirement from some more antiquated systems > and it's always a pain. It's always better to use source IP or even a trunk > prefix (dialed number prefix) which isn't really secure, but works when > security isn't an issue. > > If you have to do it that way, I'd probably turn the $Ri/$Rp into some > sort of cache key like $Ri_$Rp and map it to something like a dispatcher > setid. > > There may be a more modern way to do this, but I don't think there is. > > Good luck! > -Brett > > > On Fri, Jul 12, 2024 at 1:40 PM Sergio Charrua via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi all! >> >> for a project under development, one requirement is to be able to >> forward/relay the call to a specific destination depending on which IP >> Address and Port the call was received by Kamailio. >> This also means that Kamailio will be listening on multiple IP Address >> and Ports >> listen=udp:IP_1:port_1 # trunkgroup 1 >> listen=udp:IP_2:port_2 # trunkgroup 2 >> listen=udp:IP_3:port_3 # trunkgroup 3 >> >> I know the pv $Ri and $Rp can be used to identify from which IP address >> and port the message reached Kamailio and having those details, depending >> on the value, I can define some conditions that allows Kamailio to relay >> the call to different destination endpoints. >> For example: >> - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC >> address A.B.C.D >> - but calls from +32 to +39 received on trunkgroup2 need to go to Germany >> on SBC address 1.2.3.4 (as it is cheaper) >> >> Is there a better way of doing this without using IF/THEN/ELSE or SWITCH >> blocks and "hard code" destination SBC addresses? >> I have read the DRouting module's documentation but not sure if it could >> help and how... >> >> Any suggestions? >> >> >> >> Atenciosamente / Kind Regards / Cordialement, >> >> >> *Sérgio * >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > > > -- > > ======================================================== > Brett Nemeroff > Voice Fox Telephony LLC > Office: 512-670-8369 > Email: brett(a)voicefoxtelephony.com >

1 0

Change the To Header in the failure_route
by sadik.oualla.mohamed＠gmail.com 12 Jul '24

12 Jul '24

Hello, I need help with modifying the "To" header in the "failure_route[ID]{...}". Specifically, I want to add the "to tag" parameter from the previous SIP messages of the current dialog(e.g., 180 Ringing, PRACK or final responses). Here's the scenario: 1. Kamailio receives and relays the SIP messages: INVITE, 180 Ringing, PRACK, 200 OK, 486 Busy Here. 2. In my case, the 180 Ringing includes the "to tag" parameter. 3. When Kamailio receives the 486 response, it triggers "failure_route[ID]" using "t_on_failure(ID)" method. 4. I extract the SIP response code and reason from the final response of the INVITE transcation in this failure_route. 5. After processing and applying some instructions, I send a reply using "t_send_reply" with the code and reason extracted. The problem: - The failure_route[ID] holds the initial INVITE, which has no "to tag" parameter. - When I send a reply using "t_send_reply()", Kamailio generates a random "to tag" different from the previous SIP messages, which confuses my UAC. I think adding the "to tag" parameter manually might be a solution, but I'm not sure if I'm doing it correctly or if there is a better approach. I also tried using uac_replace_to() method, but it doesn't work in the failure_route[ID]. How can I make sure the reply in failure_route[ID] uses the correct "to tag" from the previous SIP messages? Are there any other possibilities to achieve this? Any assistance would be appreciated. Thank you!

5 11

Kemi Python3 throws a TypeError
by Michel Pelletier 12 Jul '24

12 Jul '24

Hello, Please help. I am using Kemi with the python3 module. I get random TypeError exceptions thrown within the Kamailio code. For example I have the following code: event = str(KSR.pv.get("$hdr(Event)")) log.logInfo( "KamailioStateful.ksr_request_route_loopback(): " + "Passing the request to the presence " + "publish handler for event: " + event ) rc = KSR.presence.handle_publish() if rc != 1: log.logError( "KamailioStateful.ksr_request_route_loopback(): " + "The presence publish handler returned " + "an error code: " + str(rc) ) This is what I see in the logs: 21(120) ERROR: PY3 {PUBLISH}: app_python3 [python_support.c:167]: python_handle_exception(): apy_exec: ksr_request_route((null)): Unhandled exception in the Python code: TypeError: expected bytes, str found The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 91, in ksr_request_route return ksf.ksr_request_route(msg, "Loopback") ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 100, in ksr_request_route return self.ksr_request_route_loopback(msg) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 240, in ksr_request_route_loopback rc = KSR.presence.handle_publish() ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <built-in function handle_publish> returned a result with an exception set If I put the offending line within a Try/Except it hides it but I would rather not see it not happen at all. Any Kemi Python3 users experiencing something similar? What could be causing this? Regards, Michel Pelletier

2 1

routing call depending on which IP address SIP message reached Kamailio
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! For a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

2 3

(no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! for a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

1 0

deb package build error (v5.7.6) [error: debian/rules binary subprocess returned exit status 2]
by Denys Pozniak 12 Jul '24

12 Jul '24

Hello! I'm trying to build standard deb packages for Kamailio v5.7.6 for Debian 12.6, but I get an error. By the way, there are no problems with building v5.7.2 ... #git clone --single-branch --branch 5.7.6 https://github.com/kamailio/kamailio.git /usr/src/kamailio #cd /usr/src/kamailio && ln -s pkg/kamailio/deb/bookworm debian #make cfg && RADCLI=1 make deb .... Makefile.defs defs skipped make[6]: 'libkamailio_ims.so.0.1' is up to date. gcc -shared -g -m64 -Wl,-O2 -Wl,-E -Wl,-z,relro -pthread -rdynamic "-ldl" -Wl,-Bsymbolic-functions cmd.o ims_ipsec_pcscf_mod.o ipsec.o sec_agree.o spi_gen.o spi_list.o spi_list_tests.o -lm -lmnl -L../../lib/ims/ -lkamailio_ims -Wl,-rpath,/usr/src/kamailio/src/lib/ims -o ims_ipsec_pcscf.so make[4]: Leaving directory '/usr/src/kamailio/src' make[3]: Leaving directory '/usr/src/kamailio' make[2]: Leaving directory '/usr/src/kamailio' *dpkg-buildpackage: error: debian/rules binary subprocess returned exit status 2* make[1]: Leaving directory '/usr/src/kamailio/src' -- BR, Denys Pozniak

2 3

New Member Introduction
by Maria Jonas 11 Jul '24

11 Jul '24

Hi everyone, I am new to this forum and excited to be here. I'm eager to learn more about Kamailio and its features and I'm looking forward to participating in discussions here. Could someone please guide me on how to ask questions here ? Where should I post if I have a query ? Looking forward to your advice and connecting with you all. Thanks,

2 1

usrloc ka_interval timer is not correct in kamailio 5.8
by Michal Popovič 11 Jul '24

11 Jul '24

Hello everybody, I am using kamailio in AWS behind load balancer and after upgraded from version 5.7 to 5.8 kamailio is not able to keep connection open. After quick investigation I found out that usrloc module got new parameter ka_interval that replaced timer_interval. Change is here. https://github.com/kamailio/kamailio/commit/53ace443020075f7ecd12c72f154193… Problem is that this interval is not reliable, as it is most probably still called only when timer_interval is executed. If I set values to timer_interval = 30 ka_interval = 30 ka_randomize = 20 Maximum time for keep-alive is not 30+20 seconds but 30+30+20 seconds. That is too late for AWS load balancer. I have fixed it by enabling tcp keepalives on kernel level. Is there any way how to specify exact time in version 5.8? Thanks. Michal

1 0

HTTP Client configuration path
by Sergio Charrua 11 Jul '24

11 Jul '24

Hi all! I am using HTTP_CLIENT module, and have created a http configuration file. The module's config_file parameter is set as: modparam("http_client", "config_file", "/etc/kamailio/http_client.cfg") and it works OK on my DEV environment, because there is a symlink /etc/kamailio pointing to physical folder /usr/local/etc/kamailio/ (using Kamailio 5.7.4) However in our integration environment (also using Kamailio 5.7.4), which is identical to production, and due to limitations on access rights, we cannot create a symlink /etc/kamailio pointing /usr/local/etc/kamailio/, so all cfg files are in the installation folder. So I thought that the config_file parameter should be a relative path, starting with " ./ " or by just setting the config file name, without path (as per documentation at http_client (kamailio.org) <https://kamailio.org/docs/modules/5.1.x/modules/http_client.html#idp38265140> . The issue is that even though the file http_client.cfg is present in /usr/local/etc/kamailio/, as well as other cfg files, the cfg_parser fails to load it: Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/cfg_parser.c:632]: cfg_parser_init(): cfg_parser: Unable to open file > './http_client.cfg' > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [curlcon.c:782]: http_client_load_config(): Failed to init http_client > config file parser > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [http_client.c:307]: mod_init(): Failed to load http_client connections > from [./http_client.cfg] > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/sr_module.c:1030]: init_mod(): Error while initializing module > http_client (/usr/local/lib64/kam > ailio/modules/http_client.so) I have set the parameter to > modparam("http_client", "config_file", "./http_client.cfg") and also to > modparam("http_client", "config_file", "http_client.cfg") but it only works if I add it as : > modparam("http_client", "config_file", > "/usr/local/etc/kamailio/http_client.cfg") or by adding > chroot="/usr/local/etc/kamailio" in the main cfg script, which was my final solution. Also checked for permissions, and all is fine. Note that this only seems to happen with http_client config_file parameter. I have other include commands in the main cfg file with relative paths and all works fine. I have also analysed the cfg_parser.so file, which loads the configuration file for http_client module, but could not find anything wrong with it (at 1st glance....) What am I doing wrong? Any clue? Atenciosamente / Kind Regards / Atentamente / Cordialement, *Sérgio *

1 0

SIP Question: B2BUA not ACKing 200 OK response
by Ghulam Mustafa 11 Jul '24

11 Jul '24

Hi Community, We are setting up a new provider in our VoIP setup for outgoing calls (FreeSWITCH --> Kamailio --> Provider). When the provider sends a 200 OK in response to our INVITE, FreeSWITCH's SIP stack discards the message, and we are unable to determine the cause of this behavior. I am sharing the SIP message exchange below. Could any SIP experts assist us in understanding why FreeSWITCH is not sending an ACK in response to the 200 OK message? We understand that the issue is not related to Kamailio but seems to be within the SIP stack or FreeSWITCH itself. Below are the actual INVITE and its response, which was discarded by FreeSWITCH: send 1203 bytes to tcp/[10.xxx.x.30]:5060 at 15:48:23.001054: ------------------------------------------------------------------------ INVITE sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport;branch=z9hG4bKD8p0KmvB8ZKma Max-Forwards: 69 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:gw+commsgroup@10.xxx.x.10:5080;transport=tcp;gw=commsgroup> User-Agent: FreeSWITCH-mod_sofia/1.10.11-release~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 347 X-FS-Support: update_display,send_info Remote-Party-ID: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10 >;party=calling;screen=yes;privacy=off v=0 o=Z 5395246197255529491 693195311 IN IP4 10.xxx.x.225 s=Z c=IN IP4 10.xxx.x.225 t=0 0 m=audio 55319 RTP/AVP 106 9 98 101 0 8 3 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-mux tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:8400 outgoing_send() nta: sent INVITE (85714459) to tcp/10.xxx.x.30:5060 tport.c:4191 tport_pend() tport_pend(0x150e6c0291c0): pending 0x150e6c0c52f0 for tcp/10.xxx.x.30:5060 (already 2) nua_session.c:4131 signal_call_state_change() nua(0x150d9c107460): call state changed: init -> calling, sent offer soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x150e6c0c0210, [0x150e91772798], [0x150e91772790], [(nil)]) called nua_stack.c:299 nua_stack_event() nua(0x150d9c107460): event i_state INVITE sent nua_stack.c:389 nua_application_event() nua: nua_application_event: entering nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering 2024-07-09 15:48:22.985772 100.00% [DEBUG] sofia.c:7493 Channel sofia/external/+86xxxxxxxxxxx entering state [calling][0] nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA going to sleep tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0cb390 from (tcp/10.xxx.x.30:5060) has 352 bytes, veclen = 1 recv 352 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:23.004029: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport=37555;branch=z9hG4bKD8p0KmvB8ZKma;received=10.xxx.x.10 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Server: Kamailio 5.8 Content-Length: 0 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0cb390 (352 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 100 Trying for INVITE (85714459) nta.c:3445 agent_recv_response() nta: 100 Trying is going to a transaction nta.c:9697 outgoing_estimate_delay() nta_outgoing: RTT is 3.038 ms tport.c:4253 tport_release() tport_release(0x150e6c0291c0): 0x150e6c0c52f0 by 0x150e6c0be9a0 with 0x150e6c0cb390 (preliminary) tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:5904 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x150e91973a50) nta.c:7270 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 2/3 free nta.c:1310 agent_timer() nta: timer not set tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0c9d90 from (tcp/10.xxx.x.30:5060) has 1195 bytes, veclen = 1 recv 1195 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:27.070278: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TCP 10.xxx.x.10:5080;received=10.xxx.x.10;rport=37555;branch=z9hG4bKD8p0Km From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx >;tag=7bQ0OSl9Dl-39455-57712329-1057554-14 Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:127.0.0.8;line=sr-H46V9.fbsXNDlkfEJjGbujK3J4aDo1GzP5GOJ8hcsXwV9LAWw1abHkqWlXczwL7*> Record-Route: <sip:13x.xxx.xxx.xx;transport=tcp;lr;ftag=ZSvBBtypKypDS;did=cbd.e581;nat=yes> Supported: sdp-anat Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY Require: timer Session-Expires: 2000;refresher=uas Server: Mediant VE SBC/v.7.40A.250.265 P-Asserted-Identity: <sip:4006100666@hk.chinatelecom.com;user=phone> Accept: application/sdp, application/isup, multipart/mixed, application/dtmf, application/dtmf-relay Content-Type: application/sdp Content-Length: 219 User-Agent: Kamailio 5.8 v=0 o=- 846721169 1919912032 IN IP4 13x.xxx.xxx.xx s=- c=IN IP4 13x.xxx.xxx.xx t=0 0 m=audio 49708 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:49709 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0c9d90 (1195 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 200 OK for INVITE (85714459) nta.c:3494 agent_recv_response() nta: 200 OK was discarded Any insights or suggestions would be greatly appreciated, Specifically, why is the 200 OK response being discarded?. Thank you! Regards, Mustafa

2 2

Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 11 Jul '24

11 Jul '24

Hi, Daniel. Thank you for the quick response! I will update you after testing the fix. Regards, Vanderlei

1 0

Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 10 Jul '24

10 Jul '24

Hello Henning, Thank you for your quick response. The increase happens in the core module (create-avp). See bellow: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 <---------------------------- counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 <------------------ } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } These reports above are shown after the end of calls (4000) where the outgoing INVITE is not answered. In normal calls the issue does not occur and the reports are shown below: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 430304 } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 6", "shmem:free_size = 64143184", "shmem:max_used_size = 4070816", "shmem:real_used_size = 2965680", "shmem:total_size = 67108864", "shmem:used_size = 2627856" ], "id": 683948 } Regars, Vanderlei

2 1

Intercepting OPTIONS from nathelper
by Denys Pozniak 10 Jul '24

10 Jul '24

Hello! I need advice on how to intercept local SIP OPTIONS requests from the nathelper module for some adjustments? As far as I tested, it does not fall into the *event_route[tm:local-request]* Thanks in advance! # kamailio -v version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty -- BR, Denys Pozniak

2 5

handle_subscribe(), handle_publish() and DB usage
by Володимир Іванець 04 Jul '24

04 Jul '24

Hello everyone! According to the documentation functions handle_subscribe() and handle_publish() update information not only in memory but also in the database. Isn't there a way to disable database usage for this functionality? Setting subs_db_mode to 0 does not make a change. Thank you!

1 0

Contact does not exist in PCSCF usrloc error in log
by Shantanu K Roy 03 Jul '24

03 Jul '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu Kumar Roy

3 6

Kamailio v5.6.6 Released
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, Kamailio SIP Server v5.6.6 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.6. Note that 5.6 is now the third last stable branch, v5.6.6 being the last planned released in 5.5.x series. The latest two stable branch are 5.7 and 5.8, with v5.8.2 being released a while ago. For more details about version 5.6.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-6-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

1 0

Releasing v5.6.6
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, I am considering to release Kamailio v5.6.6 (out of branch 5.6) later this week (likely on Wednesday, July 3, 2024). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. This is going to be the last release out of branch 5.6 to mark the end of official/packaging maintenance. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

Extracting and Handling Via Headers in failure_route
by sadik.oualla.mohamed＠gmail.com 02 Jul '24

02 Jul '24

Hello, I am currently working on a scenario where I need to use Kamailio in the failure_route to process SIP responses, specifically handling negative SIP replies (e.g., 486 Busy Here) that Kamailio may receive. Here is the topology: ##B2BUA (acts as a UAC with the Kamailio, because it sends the INVITE) -> Kamailio -> Callee## In my setup, Kamailio needs to extract all Via headers from the received SIP response in the failure_route, focusing particularly on the bottom Via header (from the B2BUA' UAC part). However, I am encountering difficulties in achieving this. Here is what I have tried so far: 1. To extract the Via headers, I used the following line of code in the failure_route: ```$var(viaHeaderValues) = $T_rpl($hdr(Via));``` However, this only returns the topmost Via header. 2. I attempted to get the first and last Via headers values with: ```$var(viaHeaderValueFirst) = $T_rpl($hdr(Via)[0]); $var(viaHeaderValueLast) = $T_rpl($hdr(Via)[1]);``` Unfortunately, this also only returns the topmost Via header, which contains the address of Kamailio. My goal is to make sure that when Kamailio sends the SIP reply to the UAC with send_reply method on the failure_route, it uses the extracted *Via header* of the incoming SIP response (only the bottom one). Here is an example scenario: - Received SIP Response in kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.4;branch=z9hG4bK9157.090080a91105c0ec6279bb56882d1dc8.0 Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` - Desired SIP Response from kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` The current issue is that Kamailio modifies the Via header when sending the reply with send_reply method, resulting in: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m;received=200.200.200.3 From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` Due to this change, the UAC does not correctly react to or understand the SIP response "SIP Reply follows the RFC standards, but my current UAC waits for a Via that looks like Via of the previous SIP messages in the structure of parameters.". It is crucial for the Via header to maintain the same important order such as the branch, received, and rport values as the previous exchanged SIP messages. Could you please provide guidance on how to force Kamailio to send the SIP reply using send_reply method with the extracted bottom Via header from the incoming SIP response? Any advice or examples on handling this situation would be greatly appreciated. Thank you for your time and assistance.

1 0

Kamailio v5.7.6 Released
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, Kamailio SIP Server v5.7.6 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.6. For more details about version 5.7.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-7-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.2 was released on June 12, 2024. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.7.6
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, I am considering to release Kamailio v5.7.6 (out of branch 5.7) on Tuesday, July 2, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

SRV in advertise?
by David Villasmil 01 Jul '24

01 Jul '24

Hello guys, I'm trying to use an SRV as an advertised address but kamailio fails to start. Is this not possible? Thanks, David Villasmil email: david.villasmil.work(a)gmail.com

4 7

SecSIPID Assistance
by Blake Ivey 01 Jul '24

01 Jul '24

Hi everyone. Wanting to see if someone could point me in the right direction. Still very knew to Kamailio but I am beginning to understand it better. I'm making an outbound proxy and have everything working well besides stir/shaken. I'm looking at the module page and have went back and forth with chatGPT and can't seem to figure this part out. I keep getting errors on the modparam lines. Obviously this is a self signed cert because I'm just testing. I am able to reach and download the cert from the Web server. Thank you for any assistance. # SECSIPID for Stir/Shaken modparam("secsipid", "private_key", "/etc/kamailio/secsipid/private.key") modparam("secsipid", "certificate", "/etc/kamailio/secsipid/cert.crt") modparam("secsipid", "authority_cert", "/etc/kamailio/secsipid/ca.crt") modparam("secsipid", "expire", 600) modparam("secsipid", "timeout", 2) route[STIRSHAKEN] { if (is_method("INVITE")) { if (!secsipid_add_identity("$fU", "$rU", "A", "", " http://myIPaddress.com/stir_shaken_cert.crt <http://myipaddress.com/stir_shaken_cert.crt>", "/etc/kamailio/secsipid/private.key")) { xlog("L_ERR", "Failed to sign call with ID: $ci - From: $fU\n"); send_reply("500", "Internal Server Error"); exit; } else { xlog("L_INFO", "Successfully signed call with ID: $ci - From: $fU\n"); } } # Relay the call after signing route(RELAY); }

3 8

removing Via Header with received=
by mosbah abdelkader 30 Jun '24

30 Jun '24

Hello All, I am trying to build a simple proxy that forwards SIP requests to a remote portaone server. But, kamailio is adding a Via header with received=@IP to replies got from portaone. How to disable this behavior as Asterisk (client in my setup) is not accepting this header. Thank you.

1 0

misctest module not found in kamailio debian packages
by Patrick Karton 29 Jun '24

29 Jun '24

Hello, how do i get misctest module in debian packages ? it seems not included in a package. is compilation the only way to get it ? im using kamailio 5.7.5 on debian (bullseye) 11.8 version: kamailio 5.7.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 Thanks.

1 0

Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org
by Xenofon Karamanos 28 Jun '24

28 Jun '24

Any news on regarding the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thanks, Xenofon ________________________________ From: Xenofon Karamanos via sr-dev <sr-dev(a)lists.kamailio.org> Sent: Wednesday, June 19, 2024 11:51 To: sr-dev(a)lists.kamailio.org Cc: Xenofon Karamanos <xk(a)gilawa.com> Subject: [sr-dev] Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org Hey everybody, While attempting to use the official Ubuntu package for Kamailio version 5.7.4 on Ubunut 24.04, we encountered an issue where TLS functionality does not seem to work as expected but 5.8 branch does. Given this, I am inquiring about the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thank you in advance for your assistance. Best regards, Xenofon

1 0

replies using the wrong protocol
by sarah.martin＠nextiva.com 27 Jun '24

27 Jun '24

I am extremely new at this, but trying to set up TLS with a carrier. TLS connection is good, Invite goes out, we get the 100 and the 200, but subsequent messages (ACK and BYE) are being sent with UDP and I cannot figure out how to get them to maintain the TLS transport. Any suggestions? I think this is the section I'm looking for. # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { xdbg("rtpengine_manage loop-protect MANAGE_REPLY"); #rtpengine_manage("loop-protect"); } }

6 21

Kamailio cannot forward request - invalid sendtoparameters
by serkan celik 27 Jun '24

27 Jun '24

Hi all, I face a problem when trying to forward an UPDATE request to the IMS which communicates with UDP transport protocol on one side UDP and SCTP on other side. When I try to send UPDATE towards the UDP side of IMs I see following error on kamailio side and request cannot be forwarded Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: ERROR: <core> [core/udp_server.c:597]: udp_send(): sendto(sock, buf: 0x7fc180deb5f0, len: 1509, 0, dst: (172.18.0.20:5060), tolen: 16) - err: Invalid argument (22) Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: CRITICAL: <core> [core/udp_server.c:603]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net The UPDATE message which is tried to be forwarded is as follow: Here there are double rr one is UDP and one is sctp (Route: < sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>) ) That would be great if you can help why this UPDATE cannot be forwarded. By the way double rr is on on kamailio config Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: DEBUG: <core> [core/forward.c:577]: forward_request(): Sending:#012UPDATE sip:alice@10.10.141.188;ob;sbc-id=xpllrk47odqh4pb2ofpib6vsym SIP/2.0#015#012Via: SIP/2.0/UDP 127.0.0.1:50601;branch=z9hG4bKddab.3554948ce3cfe321729448a2b082c2f3.0#015#012Via: SIP/2.0/UDP 127.0.0.1:7060;branch=z9hG4bKnHN8Q3ya#015#012To: < sip:alice@ims.wingcon.com>;tag=49IZjY2irfb#015#012From: < sip:bob@ims.wingcon.com>;tag=hfdqjj2pmui#015#012Call-Id: 3mzzecWt3Cbgvy#015#012CSeq: 1 UPDATE#015#012Max-Forwards: 70#015#012Contact: <sip:bob@10.10.141.188;ob;sbc-id=o7dpxe7qu2r34zwzio5pis2vqi>#015#012Content-Type: application/sdp#015#012Route: <sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, <sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;transport=sctp;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.120;lr>#015#012Session-Expires: 180;refresher=uas#015#012Supported: replaces, 100rel, timer, norefersub#015#012Min-SE: 90#015#012P-Charging-Vector: icid-value=e277783432fd11efba03fa163eed425f;icid-generated-at=172.18.0.120;orig-ioi=172.18.0.120;term-ioi=172.18.0.120#015#012User-Agent: PJSUA v2.14 Linux-5.4.0.172/x86_64/glibc-2.35#015#012Content-Length: 409#015#012#015#012v=0#015#012o=- 3928313948 3928313950 IN IP4 172.18.0.120#015#012s=pjmedia#015#012b=AS:84#015#012t=0 0#015#012a=X-nat:0#015#012m=audio 30106 RTP/AVP 8 96 97 121 120#015#012c=IN IP4 172.18.0.120#015#012b=TIAS:64000#015#012a=ssrc:1986855169 cname:1fca52e812ced466#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 AMR/8000#015#012a=rtpmap:97 AMR-WB/16000#015#012a=rtpmap:121 telephone-event/16000#015#012a=fmtp:121 0-16#015#012a=rtpmap:120 telephone-event/8000#015#012a=fmtp:120 0-16#015#012a=sendrecv#015#012a=rtcp:30107#015#012. Regards Serkan

1 0

Handling final response retransmissions after a reInvite
by hollis.rhonda＠gmail.com 26 Jun '24

26 Jun '24

I have a scenario where Kamailio is receiving a retransmit of a 200 OK to a late offer INVITE after it has sent the ACK, and after it has begun to handle a reInvite from the calling party. This results in the following commands to the rtpengine: Offer (reInvite), Offer (200 retransmit), Answer (ACK to retransmitted 200). At this point, I see STUN binding errors both in Chrome, where the webRTC client (called party) is running, and in the rtpengine logs. ERROR:port.cc(498)]: Received STUN BINDING error response: class=4 number=1 reason='Unauthorized' [core] STUN authentication mismatch from x.x.x.x:63396 [ice] Received ICE/STUN response code 487 for candidate pair dl2efuRG06eK4nFs:2713745946:1 from x.x.x.x::64509 to x.x.x.x [ice] ICE role change, now controlled [ice] Recalculating all ICE pair priorities [ice] Triggering check for dl2efuRG06eK4nFs:2713745946:1 [ice] ICE/STUN response with unknown transaction received (from x.x.x.x:64509 on interface x.x.x.x:16184) So, briefly, the call flow up to the point of error looks like this: INVITE w/o SDP 200 OK w/SDP, Offer to rtpengine ACK w/SDP, Answer to rtpengine reInvite w/SDP, Offer to rtpengine retransmission of 200 OK, Offer to rtpengine ACK to retransmitted 200, Answer to rtpengine I have dialog tracking enabled in my config, and it seems like Kamailio is behaving as expected when the retransmitted 200 and ACK are forwarded on, but I'm curious if other people have run into a scenario like this and how they have dealt with it.

1 0

Need help on how to implement presence_xml
by Michel Pelletier 25 Jun '24

25 Jun '24

Hello, I need to implement Event: presence, application/pidf+xml as described in RFC-3856. I understand I need to use the presence_xml module and generate PUBLISH requests to update the information and ensure that the NOTIFY get sent. I know nothing abour XCAP or wether I need it or not. Any information source out there that could help me? Regards, Michel Pelletier

1 0

TLS: client doesn't check certificate CN
by Marat Gareev 20 Jun '24

20 Jun '24

Hello, I noticed that Kamailio can route either to IP and FQDN. This means the server certificate CN is not checked by the client. How to enable something like the 'verify_peer' option? Thanks

1 0

Possibility to print values of Core Keywords in Kamailio configuration file
by sadik.oualla.mohamed＠gmail.com 20 Jun '24

20 Jun '24

Hello everyone, I have a question regarding the usage of core keywords, specifically the possibility of printing their values in log messages. While I understand that core keywords such as 'dst_port' and 'af' etc... can be accessed directly in if conditions, I am interested to know if there is a way to print the values of these core keywords for logging purposes. Thank you.

4 6

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Wed, 19 Jun 2024 11:54:05 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > What you should expect using this method is blocking while the exec runs > which could run you out of child processes while they complete. I'd also > expect to see an "unusual amount of CPU activity" which will be the result > of the forked shells being created and destroyed. > > All of that being said, yes it works. It's architecturally overly simple > and very inefficient. If your load is tiny, it might be ok. Anything > moderate like Henning and others have said, will use an inordinate amount > of cpu. I would not personally risk it. > > Thank you Brett, Henning and Alex Sadly the load won't be tiny and cannot risk it. For now I'll deploy a local nginx and fpm to run the scripts, which are in php, and will use the http_async method. Sadly we have to "http all the things" nowadays. Let's see if I'm able to write down all the logic tomorrow and do some testing. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

Adding custom headers to HTTP server response xhttp
by Sasivarunan 19 Jun '24

19 Jun '24

HI Team, I use tls and xhttp module to enable websocket interface for webrtc client to connect. I wanted to add "Strict-Transport-Security" header to xhttp response to add security. But currently I don't see a way to add it anywhere. I can see only server_header as an option, But this will get added to SIP messages as well. It 'd be great if anybody has a suggestion around this. Thanks Varunan

2 2

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