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37478 discussions

SIPP UAS scenario
by Sergio Charrua 09 May '25

09 May '25

Hi all! my DEV/LAB environment is comprised of 3 nodes: - SIP 1 with Asterisk and SIPP installed (IP 10.20.0.5) - SIP 2 with Kamailio 5.8 (IP 10.20.0.4) - SIP 3 with Asterisk and SIPP installed (IP 10.20.0.6) Kamailio is set as a stateful SIP Proxy and seems to be working fine (the configuration is pretty standard, using dispatcher to dispatch to destination, even though there is only 1 destination endpoint -» SIP 3) When establishing a call from SIP 1 using Asterisk, the termination is correctly done on SIP 3, also running Asterisk. Hanging up call on SIP 3 will send the correct BYE command to SIP 1 (of course, Kamailio is always between the dialogs) When establishing a call from SIP 1 using SIPP and UAC scenario, the termination is correctly done on SIP 3 also running SIPP with a UAS scenario. However, hanging up the call on SIP 3 (SIPP scenario has a Pause of 2 seconds then sends a BYE message), the BYE command is not forwarded by Kamailio and SIPP keep sending BYE commands. This is an example of the ACK and BYE command being sent from SIP 3: > ACK sip:10.20.0.6:5060;transport=UDP SIP/2.0 > Via: SIP/2.0/UDP > 10.20.0.4;branch=z9hG4bK680a.47d763c3f0732dbe0fdb64ccbeb3454a.0 > Via: SIP/2.0/UDP 10.20.0.5:5060 > ;received=10.20.0.5;branch=z9hG4bK01175d44;rport=5060 > Max-Forwards: 69 > From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as023237be > To: <sip:1000@10.20.0.4>;tag=20 > Contact: <sip:anonymous@10.20.0.5:5060> > Call-ID: 568f1bbe197039983fc89cc6403b927b@10.20.0.5:5060 > CSeq: 102 ACK > User-Agent: Asterisk PBX 18.17.1 > Content-Length: 0 and then: > BYE sip:anonymous@10.20.0.5:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.6:5060 > Route: <sip:10.20.0.4;lr=on;ftage=46906> > Max-Forwards: 70 > From: sipp <sip:sipp@10.20.0.6:5060>;tag=46906SIPpTag0120 > To: : "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as023237be > Call-ID: 568f1bbe197039983fc89cc6403b927b@10.20.0.5:5060 > Cseq: 2 BYE > Contact: sip:sipp@10.20.0.6:5060 > Content-Length: 0 > Subject: Sending BYE I'm pretty sure that the issue is within the SIPP Scenario(s), and my guess goes to the UAS scenario on SIP 3 that has some issue with the BYE command, but I can't figure why the message is wrong! The goal is to have a Testing environment where we can simulate different call situations. SIPP is a wonderful tool but such a PITA !! Probably i am missing some detail somewhere.... Has anyone done such a testing environment? Can anyone share a SIPP UAS scenario where the BYE command is sent from the UAS (send BYE instead of recv BYE) ? SIPP is launched on SIP 1 with: sipp -sf uac.xml -i 10.20.0.5 -p 5060 10.20.0.4 -r 1 -m 1 SIPP is launched on SIP 3 with: sipp -aa -sf uas5.xml -i 10.20.0.6 -p 5060 Thanks in advance. Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

2 2

Kamailio Crash
by Ben Kaufman 09 May '25

09 May '25

Greetings, all, I have a redirect server doing CNAM lookup (via http_async) and STIR/SHAKEN validation with secsipidx that is crashing every few hours. Kamailio 6.0.1 running in a Debian 12 container on an Amazon Linux 2023 host with ARM CPU. root@0689d4fe5fed:/# kamailio -v version: kamailio 6.0.1 (aarch64/linux) fce50d 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-NOSMP, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_SEND_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: fce50d compiled on 15:50:40 Apr 1 2025 with gcc 12.2.0 Back trace is here, so I think it's a secsipid related problem? Slightly sanitized this output (Removed SIP message specifics in step #42). What should be my next course of action? #0 __pthread_kill_implementation (threadid=281473196830752, signo=signo@entry=6, no_tid=no_tid@entry=0) at ./nptl/pthread_kill.c:44 #1 0x0000ffff95c90ab4 in __pthread_kill_internal (signo=6, threadid=<optimized out>) at ./nptl/pthread_kill.c:78 #2 0x0000ffff95c4a72c in __GI_raise (sig=sig@entry=6) at ../sysdeps/posix/raise.c:26 #3 0x0000ffff95c3747c in __GI_abort () at ./stdlib/abort.c:79 #4 0x0000ffff95c84aac in __libc_message (action=action@entry=do_abort, fmt=fmt@entry=0xffff95d66d28 "%s\n") at ../sysdeps/posix/libc_fatal.c:156 #5 0x0000ffff95c9aebc in malloc_printerr (str=str@entry=0xffff95d62098 "double free or corruption (!prev)") at ./malloc/malloc.c:5660 #6 0x0000ffff95c9cee8 in _int_free (av=0xffff95db0af0 <main_arena>, p=p@entry=0xaaab05dd41b0, have_lock=<optimized out>, have_lock@entry=0) at ./malloc/malloc.c:4587 #7 0x0000ffff95c9f77c in __GI___libc_free (mem=<optimized out>) at ./malloc/malloc.c:3385 #8 0x0000ffff90eb0a0c in w_secsipid_get_url (msg=0xffff92f53030, purl=0xffff92e84e10 "\220M\350\222\377\377", povar=0xffff92e84fd0 "\004") at secsipid_mod.c:961 #9 0x0000aaaaea4fd710 in do_action (h=0xffffff40a610, a=0xffff92e8e9b0, msg=0xffff92f53030) at core/action.c:1133 #10 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40a610, a=0xffff92e8e9b0, msg=0xffff92f53030) at core/action.c:1620 #11 0x0000aaaaea50db44 in run_actions_safe (h=0xffffff40bc50, a=0xffff92e8e9b0, msg=0xffff92f53030) at core/action.c:1683 #12 0x0000aaaaea71a7e0 in rval_get_long (h=0xffffff40bc50, msg=0xffff92f53030, i=0xffffff40ab48, rv=0xffff92e8fc78, cache=0x0) at core/rvalue.c:973 #13 0x0000aaaaea7200f0 in rval_expr_eval_long (h=0xffffff40bc50, msg=0xffff92f53030, res=0xffffff40ab48, rve=0xffff92e8fc70) at core/rvalue.c:1852 #14 0x0000aaaaea720118 in rval_expr_eval_long (h=0xffffff40bc50, msg=0xffff92f53030, res=0xffffff40b080, rve=0xffff92e903b0) at core/rvalue.c:1862 #15 0x0000aaaaea4fd090 in do_action (h=0xffffff40bc50, a=0xffff92e91810, msg=0xffff92f53030) at core/action.c:1099 #16 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40bc50, a=0xffff92e8d690, msg=0xffff92f53030) at core/action.c:1620 #17 0x0000aaaaea4f9400 in do_action (h=0xffffff40bc50, a=0xffff92e82210, msg=0xffff92f53030) at core/action.c:711 #18 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40bc50, a=0xffff92e82210, msg=0xffff92f53030) at core/action.c:1620 #19 0x0000aaaaea50db44 in run_actions_safe (h=0xffffff40daf0, a=0xffff92e82210, msg=0xffff92f53030) at core/action.c:1683 #20 0x0000aaaaea71a7e0 in rval_get_long (h=0xffffff40daf0, msg=0xffff92f53030, i=0xffffff40c188, rv=0xffff92e82378, cache=0x0) at core/rvalue.c:973 #21 0x0000aaaaea7200f0 in rval_expr_eval_long (h=0xffffff40daf0, msg=0xffff92f53030, res=0xffffff40c188, rve=0xffff92e82370) at core/rvalue.c:1852 #22 0x0000aaaaea720118 in rval_expr_eval_long (h=0xffffff40daf0, msg=0xffff92f53030, res=0xffffff40c6c0, rve=0xffff92e82ab0) at core/rvalue.c:1862 #23 0x0000aaaaea4fd090 in do_action (h=0xffffff40daf0, a=0xffff92e83a90, msg=0xffff92f53030) at core/action.c:1099 #24 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40daf0, a=0xffff92e83a90, msg=0xffff92f53030) at core/action.c:1620 #25 0x0000aaaaea4fd5e8 in do_action (h=0xffffff40daf0, a=0xffff92e83bf0, msg=0xffff92f53030) at core/action.c:1119 #26 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40daf0, a=0xffff92e79450, msg=0xffff92f53030) at core/action.c:1620 #27 0x0000aaaaea4f9400 in do_action (h=0xffffff40daf0, a=0xffff92e75710, msg=0xffff92f53030) at core/action.c:711 #28 0x0000aaaaea50d3c4 in run_actions (h=0xffffff40daf0, a=0xffff92e75710, msg=0xffff92f53030) at core/action.c:1620 #29 0x0000aaaaea50db44 in run_actions_safe (h=0xffffff4101f8, a=0xffff92e75710, msg=0xffff92f53030) at core/action.c:1683 #30 0x0000aaaaea71a7e0 in rval_get_long (h=0xffffff4101f8, msg=0xffff92f53030, i=0xffffff40e028, rv=0xffff92e75878, cache=0x0) at core/rvalue.c:973 #31 0x0000aaaaea7200f0 in rval_expr_eval_long (h=0xffffff4101f8, msg=0xffff92f53030, res=0xffffff40e028, rve=0xffff92e75870) at core/rvalue.c:1852 #32 0x0000aaaaea720118 in rval_expr_eval_long (h=0xffffff4101f8, msg=0xffff92f53030, res=0xffffff40e560, rve=0xffff92e75fb0) at core/rvalue.c:1862 #33 0x0000aaaaea4fd090 in do_action (h=0xffffff4101f8, a=0xffff92e76e10, msg=0xffff92f53030) at core/action.c:1099 #34 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92e74500, msg=0xffff92f53030) at core/action.c:1620 #35 0x0000aaaaea5095fc in do_action (h=0xffffff4101f8, a=0xffff92e79110, msg=0xffff92f53030) at core/action.c:1401 #36 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92e71a40, msg=0xffff92f53030) at core/action.c:1620 #37 0x0000aaaaea4f9400 in do_action (h=0xffffff4101f8, a=0xffff92dd8520, msg=0xffff92f53030) at core/action.c:711 #38 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92dd7e30, msg=0xffff92f53030) at core/action.c:1620 #39 0x0000aaaaea4fd5a8 in do_action (h=0xffffff4101f8, a=0xffff92dd8680, msg=0xffff92f53030) at core/action.c:1115 #40 0x0000aaaaea50d3c4 in run_actions (h=0xffffff4101f8, a=0xffff92dce4c0, msg=0xffff92f53030) at core/action.c:1620 #41 0x0000aaaaea50dbf4 in run_top_route (a=0xffff92dce4c0, msg=0xffff92f53030, c=0x0) at core/action.c:1703 #42 0x0000aaaaea6bfff8 in receive_msg ( buf=0xaaaaeac284b8 <buf> "INVITE sip:2345678901@1.2.3.4 SIP/2.0\r\nRecord-Route: <sip:1.2.3.5;lr=on;ftag=gK045643d5>\r\nRecord-Route: <sip:1.2.3.6;lr=on;ftag=gK045643d5>\r\nVia: SIP/2.0/UDP 1.2.3.5:5060;branch"..., len=2492, rcv_info=0xffffff410a10) at core/receive.c:520 #43 0x0000aaaaea85abe0 in udp_rcv_loop () at core/udp_server.c:770 #44 0x0000aaaaea4e26a8 in main_loop () at main.c:1895 #45 0x0000aaaaea4f1e88 in main (argc=16, argv=0xffffff411178) at main.c:3406 Kaufman Senior Voice Engineer E: bkaufman(a)bcmone.com SIP.US Client Support: 800.566.9810 | SIPTRUNK Client Support: 800.250.6510 | Flowroute Client Support: 855.356.9768 [img]<https://www.sip.us> [img]<https://www.siptrunk.com> [img]<https://www.flowroute.com> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

2 3

Having issues with compilation of evapi module and evapi_async_relay functions
by Steve Lim 02 May '25

02 May '25

Hello, We're trying to compile the evapi module. Tested on an upgraded 5.8.6 install as well as a new 6.0.1 install. tm module is loaded (for async), but our issues seem to run deeper. evapi_relay and evapi_async_relay are both not found by the core loader as per load failure (same pattern for evapi_async_relay, ommited for brevity) : 0(21496) DEBUG: <core> [core/sr_module.c:766]: find_mod_export_record(): export of <evapi_relay> not found (flags 1) 0(21496) DEBUG: <core> [core/sr_module.c:766]: find_mod_export_record(): export of <evapi_relay> not found (flags 0) 0(21496) ERROR: <core> [core/cfg.y:3877]: yyparse(): cfg. parser: failed to find command evapi_relay (params 2) 0(21496) CRITICAL: <core> [core/cfg.y:4018]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 111, column 43: unknown command, missing loadmodule? earlier up, we're able to set the bind_addr parameter, so it appears that kamailio does recognize the module (and would fail anyways if it didn't since it's supposed to be loaded) 0(21496) DEBUG: <core> [core/modparam.c:112]: set_mod_param_regex(): 'evapi' matches module 'evapi' 0(21496) DEBUG: <core> [core/sr_module.c:852]: find_param_export(): found <bind_addr> in module evapi [/usr/local/lib64/kamailio/modules/evapi.so] 0(21496) DEBUG: <core> [core/modparam.c:135]: set_mod_param_regex(): found <bind_addr> in module evapi [/usr/local/lib64/kamailio/modules/evapi.so] quick look with nm shows that evapi_relay is exported, but evapi_async_relay isn't. root@kamailio-devel:~# nm -D /usr/local/lib64/kamailio/modules/evapi.so | grep relay 0000000000011dd7 T _evapi_relay 0000000000013b3f T evapi_relay 0000000000013b63 T evapi_relay_multicast 0000000000013b8d T evapi_relay_unicast What am I missing out here? Why can't I we get to the evapi_relay exported function and why isn't evapi_async_relay even showing up as exported? Thanks!

1 1

Kamailio Asterisk Realtime integration
by cedric＠mrtn.ovh 02 May '25

02 May '25

Hello, I'm trying to use Kamailio as proxy for Asterisk. I follow this tutorial https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I maked some updates to replace old libs and old elements related to db table and fields to use the pjsip table (ps_auths) and fields (username, password). I configured Asterisk Realtime by following the Asterisk documentation (extconfig.conf, sorcery.conf, rec_odbc.conf etc...). I'm lock at this step : The registering is handle correctly by Kamailio and forwarded to Asterisk, but asterisk respond with a "401" error and want to get a digest auth... Bellow, my kamailio config file: ``` #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_DEBUG # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asterisk_password@localhost/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.1.2:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.2" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "192.168.1.2" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "jsonrpcs.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs fifo params ----- modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "ps_auths", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif ``` Also, I'm not sure about the configuration to add into asterisk, like a kamailio registration block or endpoint block, and... probably others... Thanks to help. Flywall.

1 0

WebRTC error NS_ERROR_NET_TIMEOUT
by David Cunningham 01 May '25

01 May '25

Hello, Does anyone know of a reason (besides the obvious like network down) why users might experience a NS_ERROR_NET_TIMEOUT error in their browser when connecting to WebRTC port 8443 on Kamailio (version 5.2.7)? This only happens intermittently, so debugging it is tricky. The number of connections to Kamailio port 8443 peaks at around 4150 daily. The max open files ulimit in Linux is 65535, and in the Kamailio configuration we have "tcp_max_connections=32768" and "tls_max_connections=32000". Would anyone have any suggestions on where the problem might be? Thanks in advance! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

Looking for Documentation/Guidance to Setup Kamailio and RTPProxy to Terminate Encryption for Asterisk
by Joshua Ruehlig 30 Apr '25

30 Apr '25

Hello, I am fairly new to SIP and just recently setup Asterisk for a few purposes. 1) To have my Hikvision doorbell videocall my phone when someone presses it 2) Videocalling between me and my families' phones/tablets 3) Eventually have a home phone setup for POTS calling. I currently have an ObiHai202 that is still working with GoogleVoice, so this not a priority at the moment. I wanted to have encryption for SIP and RTP outside of my network, but wanted this to all be terminated on my OPNsense router. I can install Kamailio and RTPProxy on OPNsense since they are available for FreeBSD, but it looks like Rtpengine is currently not ported. Ideally I wanted most logic to be handled on my Asterisk server, and have Kamailio/RTPProxy do the bare minimum of encrypting/decryption and rewriting headers as needed. ## Diagram of my Targeted Setup ## * Android Phones / Tablets running Linphone || SIPS(TLS) / SRTP || * OPNsense Router ~ WAN_IP(dynamic)=DOMAIN.com / LAN_IP=10.0.1.1 - Kamailio / RTPProxy || SIP(UDP) / RTP || * FreeBSD Application Server ~ LAN_IP=10.0.1.17 / Internal_IP=192.168.10.17 - Asterisk || SIP(UDP) / RTP || * Hikvision Doorbell ~ Internal_IP=192.168.10.100 #### From my reading so far I have a few questions... * To setup Kamailio > Asterisk it looks like I need to setup the dispatcher module; I also I may need to write some routing code. Am I correct, that I will need to use the dispatcher module? Will I need to modify the routing code/logic or is the default usually sufficient? * I read RTPProxy does support SRTP, but I don't see any configuration options related to encryption. I assume this is just enabled in Asterisk/Kamailio? * For Asterisk configuration I believe I just need to define the outbound_proxy. But now I am not sure how I can require SRTP between my Linphone clients and RTPProxy, while the leg between RTPProxy and Asterisk is unencrypted RTP? Thanks for any help that is provided!

1 2

Kamailio TLS Listener Not Binding to Port (5061 or 10061)
by steven.muchwe 29 Apr '25

29 Apr '25

Hello Kamailio Community, I’ve been troubleshooting an issue with Kamailio where the TLS listener fails to bind to any specified ports (5061, 10061, etc.), despite valid configurations and certificates. Here’s a summary of my setup and steps taken: 1. *Environment Details*: - Kamailio version: 5.7.4 - OpenSSL version: 3.0.13 - Operating System: Ubuntu (Noble Release) - TLS module (tls.so) is installed and loaded. 2. *Issue Details*: - Configurations validate successfully (config file ok). - OpenSSL works perfectly when testing certificates and keys with s_server, binding to ports (5061 and others). - Kamailio fails to bind TLS listeners (ss -tulnp shows no activity on the specified ports). 3. *Steps Already Taken*: - Simplified TLS configuration (minimal_tls.cfg) with: plaintext listen=tls:10.14.202.39:5061 loadmodule "tls.so" modparam("tls", "certificate", "/home/localtech/vicissl/868a963bc33d5eae.crt") modparam("tls", "private_key", "/home/localtech/vicissl/private.key") - Tested multiple ports (5061, 10061, 15061). - Checked firewall settings (iptables) and confirmed no restrictions. - Rebuilt Kamailio from source and ensured TLS modules are linked to OpenSSL. - Ran Kamailio with maximum debugging (-ddd) to examine logs—no binding-related errors appeared. 4. *Log Excerpts*: (Attach relevant logs showing TLS initialization or lack of binding activity.) 5. *Question*: What additional steps or configurations should I explore to resolve this issue? Could this be a compatibility problem between Kamailio 5.7.4 and OpenSSL 3.0.13? Any guidance or insights would be greatly appreciated! Best regards, Steven Muchwe Njoroge

4 8

Re: SIP Call Drops After 0.32 Seconds – Asterisk on Kubernetes with Kamailio + RTPengine
by airsay＠duck.com 29 Apr '25

29 Apr '25

I will let the more experienced Kamailio folks comment on the technical/Kamailio/RTP modifications that you need to make. The SIP standard requires an ACK to a 200 OK to be sent within that 32 second window. If an ACK is not received, the session is torn down. You would need to do a packet capture to review the Contact header in the SIP and the c: in the SDP. I'm just getting bedded in with SIP, reading Alan B Johnston's "SIP: Understanding the Session Initiation Protocol". Excellent introduction to understanding SIP. Best regards Sent from my iPhone > On 23 Apr 2025, at 16:35, Fernando Lopes via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello everyone, > I'm running into an issue with SIP calls in my current setup and would really appreciate some help. > > Setup: > I have a machine named sip00 (IP: 192.168.1.75) running Kamailio + RTPengine. > Kamailio is dispatching calls to sip:192.168.1.190:32210;transport=tcp. This IP points to another machine running Asterisk inside a Kubernetes cluster. > RTPengine is configured with an RTP port range of 10000–20000, and my router is set to allow that same range. > > Asterisk Kubernetes Service Configuration: > yaml > Copy > Edit > spec: > ports: > - name: tcp-port > protocol: TCP > port: 5060 > targetPort: 5060 > nodePort: 32210 > - name: udp-port > protocol: UDP > port: 5060 > targetPort: 5060 > nodePort: 32210 > > Problem: > When I initiate a SIP call, the router forwards traffic to Kamailio + RTPengine, which then sends it to the Asterisk server on 192.168.1.190. > Everything seems fine initially, but at exactly 0.32 seconds into the call, Asterisk sends a BYE and no longer responds with 200 OK to the SIP dialog — even though I'm still receiving and sending audio. Then, at around 01:04, I get a 408 Request Timeout. > > Questions: > Do I need to explicitly expose the RTP port range (10000–20000) in the Asterisk Kubernetes service as well? > Why is Asterisk sending a BYE so early if audio is still flowing? > Could it be a signaling timeout or an issue with SIP dialog tracking? > Any help or pointers would be greatly appreciated! > > Thanks in advance! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users(a)lists.kamailio.org > 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!

7 7

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

SIP Call Drops After 0.32 Seconds – Asterisk on Kubernetes with Kamailio + RTPengine
by Fernando Lopes 29 Apr '25

29 Apr '25

Hello everyone, I'm running into an issue with SIP calls in my current setup and would really appreciate some help. Setup: I have a machine named sip00 (IP: 192.168.1.75) running Kamailio + RTPengine. Kamailio is dispatching calls to sip:192.168.1.190:32210;transport=tcp. This IP points to another machine running Asterisk inside a Kubernetes cluster. RTPengine is configured with an RTP port range of 10000–20000, and my router is set to allow that same range. Asterisk Kubernetes Service Configuration: yaml Copy Edit spec: ports: - name: tcp-port protocol: TCP port: 5060 targetPort: 5060 nodePort: 32210 - name: udp-port protocol: UDP port: 5060 targetPort: 5060 nodePort: 32210 Problem: When I initiate a SIP call, the router forwards traffic to Kamailio + RTPengine, which then sends it to the Asterisk server on 192.168.1.190. Everything seems fine initially, but at exactly 0.32 seconds into the call, Asterisk sends a BYE and no longer responds with 200 OK to the SIP dialog — even though I'm still receiving and sending audio. Then, at around 01:04, I get a 408 Request Timeout. Questions: Do I need to explicitly expose the RTP port range (10000–20000) in the Asterisk Kubernetes service as well? Why is Asterisk sending a BYE so early if audio is still flowing? Could it be a signaling timeout or an issue with SIP dialog tracking? Any help or pointers would be greatly appreciated! Thanks in advance!

2 1

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