- sr-users - kamailio.org

kamailio 4.4 writes wrong expires parameter at the database
by Yuriy Gorlichenko 27 Jan '16

27 Jan '16

2 3

Fwd: Invalid Configuration File
by Andrew J. Smith 27 Jan '16

27 Jan '16

Hello! I followed this tutorial (http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour) to the T using virtual box and a debian wheezy image. Actually, I did change one thing: This repository "deb http://deb.kamailio.org/kamailio31-nightly squeeze main" always resulted in a file not found error. So I used these two lines instead: deb http://deb.kamailio.org/kamailio squeeze main deb-src http://deb.kamailio.org/kamailio squeeze main Everything else is fine except for the fact that the MySQL users have different names/passwords. I was able to generate users and configured everything as posted. When I start the service (/etc/init.d/kamailio start) I get an error message: "Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed!" followed by a looooooong list of errors (copied below). I retried with Kamailio33 (31 is not available as far as I can tell). This version has the MySQL users indicated in the tutorial (openser, openserro) - newer versions have a MySQL user named kamailio... Anyway, same long list of errors as before. I also tried the same procedure on a Raspberry Pi... Everything goes exactly as described until I try to start the Kamailio server. If I don't replace the original kamailio.cfg file with the one linked to in the tutorial, the server starts, but I can't seem to connect to it using Jitsi. What am I doing wrong? Any help would be greatly appreciated. Best regards, Andrew -------------------------------------------------------- 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <db_mysql> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 191, column 12-24: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <mi_fifo> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 194, column 12-23: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <kex> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 195, column 12-19: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <tm> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 196, column 12-18: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <tmx> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 197, column 12-19: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <sl> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 198, column 12-18: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <rr> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 199, column 12-18: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <pv> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 200, column 12-18: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 201, column 12-22: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <usrloc> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 202, column 12-22: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <registrar> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 203, column 12-25: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <textops> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 204, column 12-23: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <siputils> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 205, column 12-24: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <xlog> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 206, column 12-20: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <sanity> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 207, column 12-22: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <ctl> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 208, column 12-19: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <mi_rpc> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 209, column 12-22: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <acc> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 210, column 12-19: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <auth> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 213, column 12-20: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <auth_db> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 214, column 12-23: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <nathelper> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 234, column 12-25: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <rtpproxy> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 235, column 12-24: failed to load module 0(20143) ERROR: <core> [sr_module.c:565]: load_module(): could not find module <tls> in </usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/> 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 239, column 12-19: failed to load module 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <mi_fifo> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 255, column 54: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <tm> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 260, column 39: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <tm> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 262, column 33: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <tm> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 264, column 38: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rr> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 269, column 35: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rr> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 271, column 35: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <registrar> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 275, column 44: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 284, column 33: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 285, column 32: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 286, column 36: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 290, column 38: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 292, column 30: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 293, column 37: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 295, column 73: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 296, column 45: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <usrloc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 310, column 75: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <usrloc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 311, column 32: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <usrloc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 312, column 35: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <auth_db> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 318, column 76: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <auth_db> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 319, column 41: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <auth_db> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 320, column 50: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <auth_db> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 321, column 43: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <auth_db> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 322, column 36: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpproxy> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 363, column 77: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <nathelper> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 366, column 45: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <nathelper> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 367, column 43: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <nathelper> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 368, column 41: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <nathelper> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 369, column 64: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <nathelper|registrar> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 372, column 65: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <usrloc> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 373, column 34: Can't set module parameter 0(20143) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <tls> found 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 379, column 50: Can't set module parameter 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command is_method (params 1) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 418, column 24: unknown command, missing loadmodule? 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command t_check_trans (params 0) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 420, column 21: unknown command, missing loadmodule? 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command t_relay (params 0) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 421, column 12: unknown command, missing loadmodule? 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command t_check_trans (params 0) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 425, column 16: unknown command, missing loadmodule? 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command remove_hf (params 1) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 432, column 19: unknown command, missing loadmodule? 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command is_method (params 1) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 433, column 34: unknown command, missing loadmodule? 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command record_route (params 0) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 434, column 16: unknown command, missing loadmodule? 0(20143) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command is_method (params 1) 0(20143) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 437, column 24: unknown command, missing loadmodule? 0(20143) ERROR: <core> [pvapi.c:825]: pv_parse_spec2(): error searching pvar "rU" 0(20143) ERROR: <core> [pvapi.c:1029]: pv_parse_spec2(): wrong char [U/85] in [$rU] at [2 (0)] 0(20143) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 453, column 6-8: Can't get from cache: $rU ERROR: bad config file (63 errors) 0(20143) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized ... failed! failed! --------------------------------------------------------

2 1

doubt using sip tcp creating new transaction
by david escartin 27 Jan '16

27 Jan '16

hello all i'm facing some weird log from kamailio (i think they are weird) when using sip tcp in the caller side and udp in the callee side. seems like the tcp socket is active in the caller side and the call is connected, since the invite transaction completes. After that, if we receive an in-dialog request from the callee side, the kamailio doesnt find the tcp connection created and it has to create again the socket by SYN procedure for the other conn way. up to this point i think it's everything correct. A----the thing i dont understand, is that checking version 4.2.6, the logs i have when the request in-dialog comes from UAS, are like these 5(2979) DEBUG: <core> [tcp_main.c:1820]: tcp_send(): tcp_send: no open tcp connection found, opening new one 5(2979) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 79.170.68.171 5(2979) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 5063, type 2 5(2979) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 1522:2178:0, 4 5(2979) DEBUG: <core> [tcp_main.c:2699]: tcpconn_1st_send(): pending write on new connection 0x7fac1168f028 (-1/968 bytes written) 5(2979) DEBUG: tm [t_funcs.c:395]: t_relay_to(): SER: new transaction fwd'ed B----while when using 4.2.2 or 4.4 1(791) DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one 1(791) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 79.170.68.171 1(791) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 5063, type 2 1(791) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 1522:3421:0, 4 1(791) INFO: <core> [tcp_main.c:2753]: tcpconn_1st_send(): quick connect for 0x7f880540f758 1(791) DEBUG: tm [t_funcs.c:394]: t_relay_to(): SER: new transaction fwd'ed the difference between A and B is that in B i use dialog flags to do the t_relay_to_tcp for the indialog requests (and not in A), and in A i use the advertised IP in the listen addresses since kamailio is behind a NAT, while B machine scenario has public IPs. could those 2 things explain the ebhaviour difference? is there anything abnormal in the case B? thanks a lot and regards david escartin

3 5

Re: [SR-Users] Kamailio IMS deployment
by sainath.ellendula＠wipro.com 27 Jan '16

27 Jan '16

Hi Franz, Thanks for the response. As suggested followed below commands to check the GRANTs on database hss_db. mysql -u root -p show grants for 'hss'@'server.net1.test'; The output result is same as mentioned below. Could you please guide me here to proceed further. Regards, Sainath ---------- Forwarded message ---------- From: "Franz Edler" <franz.edler(a)technikum-wien.at<mailto:franz.edler@technikum-wien.at>> Date: Jan 27, 2016 1:07 AM Subject: RE: [SR-Users] Kamailio IMS deployment To: "Kranti Kumar" <krantikumara(a)gmail.com<mailto:krantikumara@gmail.com>>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Cc: "Ramya Y" <y.ramya146(a)gmail.com<mailto:y.ramya146@gmail.com>> Obviously the access of HSS to the database did not succeed. You should check if access permissions to the database is enabled for HSS. Check the GRANTs on the database hss_db. mysql -u root -p show grants for 'hss'@'server.net1.test'; quit; The result should be: +-------------------------------------------------------------------------------------------------------------------+ | Grants for hss(a)server.net1.test<mailto:hss@server.net1.test> | +-------------------------------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO 'hss'@'server.net1.test' IDENTIFIED BY PASSWORD '*AC83FCF24B79020EF2C51D780CDFA559F25CC9BE' | | GRANT SELECT, INSERT, UPDATE, DELETE ON `hss_db`.* TO 'hss'@'server.net1.test' | +-------------------------------------------------------------------------------------------------------------------+ BR Franz From: Kranti Kumar [mailto:krantikumara@gmail.com<mailto:krantikumara@gmail.com>] Sent: Sunday, January 24, 2016 10:55 AM To: franz.edler(a)technikum-wien.at<mailto:franz.edler@technikum-wien.at>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Cc: Ramya Y <y.ramya146(a)gmail.com<mailto:y.ramya146@gmail.com>> Subject: Re: [SR-Users] Kamailio IMS deployment Thanks a lot Franz. Observed few issues even after changes done in /etc/bind. Please find the below Errors in each interfaces and also find the attachment of /var/log/syslog(not shared full logs) Could you please help me here to proceed further. HSS - Configured IP : 10.0.0.9 root@server:~# ./hss.sh - Configured IP : 10.0.0.9 Building Classpath Classpath is lib/xml-apis.jar:lib/xercesImpl.jar:lib/xerces-2.4.0.jar:lib/xalan-2.4.0.jar:lib/tomcat-util.jar:lib/tomcat-http.jar:lib/tomcat-coyote.jar:lib/struts.jar:lib/servlets-default.jar:lib/servlet-api.jar:lib/naming-resources.jar:lib/naming-factory.jar:lib/mysql-connector-java-3.1.12-bin.jar:lib/mx4j-3.0.1.jar:lib/log4j.jar:lib/junit.jar:lib/junitee.jar:lib/jta.jar:lib/jsp-api.jar:lib/jmx.jar:lib/jdp.jar:lib/jasper-runtime.jar:lib/jasper-compiler-jdt.jar:lib/jasper-compiler.jar:lib/hibernate3.jar:lib/FHoSS.jar:lib/ehcache-1.1.jar:lib/dom4j-1.6.1.jar:lib/commons-validator.jar:lib/commons-modeler.jar:lib/commons-logging.jar:lib/commons-logging-1.0.4.jar:lib/commons-lang.jar:lib/commons-fileupload.jar:lib/commons-el.jar:lib/commons-digester.jar:lib/commons-collections-3.1.jar:lib/commons-beanutils.jar:lib/cglib-2.1.3.jar:lib/catalina-optional.jar:lib/catalina.jar:lib/c3p0-0.9.1.jar:lib/base64.jar:lib/asm.jar:lib/asm-attrs.jar:lib/antlr-2.7.6.jar::log4j.properties:.. 2016-01-21 14:07:07,876 INFO de.fhg.fokus.hss.main.TomcatServer - startTomcat Tomcat-Server is started. 2016-01-21 14:07:12,462 WARN org.apache.catalina.connector.MapperListener - registerEngine Unknown default host: 10.0.0.9 2016-01-21 14:07:16,578 INFO de.fhg.fokus.hss.web.servlet.ResponseFilter - init Response Filter Initialisation! 2016-01-21 14:07:17,966 INFO de.fhg.fokus.hss.main.TomcatServer - startTomcat WebConsole of FHoSS was started ! 2016-01-21 14:08:21,200 WARN com.mchange.v2.resourcepool.BasicResourcePool - run com.mchange.v2.resourcepool.BasicResourcePool$AcquireTask@1670988b<about:blank> -- Acquisition Attempt Failed!!! Clearing pending acquires. While trying to acquire a needed new resource, we failed to succeed more than the maximum number of allowed acquisition attempts (60). Last acquisition attempt exception: com.mysql.jdbc.CommunicationsException: Communications link failure due to underlying exception: ** BEGIN NESTED EXCEPTION ** java.net.SocketException MESSAGE: java.net.ConnectException: Connection refused STACKTRACE: java.net.SocketException: java.net.ConnectException: Connection refused PCSCF - Configured IP : 10.0.0.10 root@server:~# kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2352) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: pcscf.net1.test [10.0.0.10]:4060 Aliases: *: pcscf.net1.test:* WARNING: no fork mode 0(2352) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(2352) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(2352) INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 0(2352) INFO: <core> [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(2352) INFO: <core> [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(2352) WARNING: ims_usrloc_pcscf [usrloc_db.c:59]: connect_db(): DB connection already open... continuing ^C^C 0(2352) NOTICE: <core> [main.c:683]: handle_sigs(): Thank you for flying kamailio!!! 1(2373) INFO: <core> [main.c:798]: sig_usr(): signal 15 received 2(2374) INFO: <core> [main.c:798]: sig_usr(): signal 15 received 0(2352) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized ICSCF - Configured IP : 10.0.0.11 root@server:~# kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2353) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: icscf.net1.test [10.0.0.11]:5060 Aliases: *: icscf.net1.test:* WARNING: no fork mode 0(2353) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(2353) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(2353) INFO: <core> [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(2353) INFO: <core> [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(2353) INFO: cdp [mod.c:227]: cdp_child_init(): CDiameterPeer child starting ... 0(2353) INFO: cdp [mod.c:229]: cdp_child_init(): ... CDiameterPeer child started 10(2370) INFO: cdp [timer.c:205]: timer_process(): Timer process starting up... 6(2366) INFO: cdp [worker.c:331]: worker_process(): [3] Worker process started... 7(2367) INFO: cdp [receiver.c:445]: receiver_process(): receiver_process(): [] Receiver process doing init on new process... 7(2367) INFO: cdp [receiver.c:450]: receiver_process(): receiver_process(): [] Receiver process starting up... 8(2368) INFO: cdp [receiver.c:445]: receiver_process(): receiver_process(): [hss.net1.test] Receiver process doing init on new process... 8(2368) INFO: cdp [receiver.c:183]: add_serviced_peer(): add_serviced_peer(): Adding serviced_peer_t to receiver for peer [hss.net1.test] 9(2369) INFO: cdp [acceptor.c:82]: acceptor_process(): Acceptor process starting up... 9(2369) WARNING: cdp [tcp_accept.c:123]: create_socket(): create_socket(): Trying to open/bind/listen on 10.0.0.9 port 3869 9(2369) WARNING: cdp [tcp_accept.c:148]: create_socket(): create_socket(): Successful socket open/bind/listen on 10.0.0.9 port 3869 9(2369) INFO: cdp [acceptor.c:96]: acceptor_process(): Acceptor opened sockets. Entering accept loop ... 8(2368) INFO: cdp [receiver.c:450]: receiver_process(): receiver_process(): [hss.net1.test] Receiver process starting up... 10(2370) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 0 10(2370) INFO: cdp [peerstatemachine.c:526]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer hss.net1.test 10(2370) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 10.0.0.9 port 3868 10(2370) WARNING: cdp [receiver.c:911]: peer_connect(): peer_connect(): Error opening connection to to 10.0.0.9 port 3868 >Connection refused 10(2370) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 0 10(2370) INFO: cdp [peerstatemachine.c:526]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer hss.net1.test 10(2370) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 10.0.0.9 port 3868 10(2370) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.net1.test:3868 connected 10(2370) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 6 10(2370) WARNING: cdp [peermanager.c:315]: peer_timer(): Inactivity on peer [hss.net1.test], sending DWR... - if we don't get a reply, the peer will be closed 8(2368) INFO: cdp [receiver.c:802]: receive_loop(): select_recv(): [hss.net1.test] read on socket [14] returned 0 > ... dropping 8(2368) INFO: cdp [receiver.c:219]: disconnect_serviced_peer(): drop_serviced_peer(): [hss.net1.test] Disconnecting from peer ^C10(2370) INFO: <core> [main.c:798]: sig_usr(): signal 2 received 0(2353) ALERT: <core> [main.c:725]: handle_sigs(): child process 2361 exited normally, status=0 0(2353) INFO: <core> [main.c:740]: handle_sigs(): dont_fork turned on, living on 0(2353) NOTICE: <core> [main.c:683]: handle_sigs(): Thank you for flying kamailio!!! 2(2362) INFO: <core> [main.c:798]: sig_usr(): signal 15 received 0(2353) INFO: cdp [mod.c:242]: cdp_exit(): CDiameterPeer child stopping ... 0(2353) INFO: cdp [diameter_peer.c:428]: diameter_peer_destroy(): destroy_diameter_peer(): Terminating all children... 0(2353) INFO: cdp [diameter_peer.c:435]: diameter_peer_destroy(): destroy_diameter_peer(): Waiting for child [2370] to terminate... 0(2353) INFO: cdp [diameter_peer.c:447]: diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated. Cleaning up. 0(2353) INFO: cdp [worker.c:140]: worker_destroy(): Unlocking workers waiting on empty queue... 0(2353) INFO: cdp [worker.c:143]: worker_destroy(): Unlocking workers waiting on full queue... 0(2353) CRITICAL: cdp [diameter_peer.c:487]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test 0(2353) INFO: cdp [mod.c:244]: cdp_exit(): ... CDiameterPeer child stoped 0(2353) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized SCSCF - Configured IP : 10.0.0.12 root@server:/etc# kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2371) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: scscf.net1.test [10.0.0.12]:6060 Aliases: *: scscf.net1.test:* WARNING: no fork mode 0(2371) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(2371) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(2371) INFO: ims_usrloc_scscf [hslot.c:69]: ul_init_locks(): locks array size 512 0(2371) INFO: ims_usrloc_scscf [hslot_sp.c:67]: subs_init_locks(): locks array size 512 0(2371) INFO: ims_usrloc_scscf [contact_hslot.c:67]: init_contacts_locks(): locks array size 512 0(2371) INFO: <core> [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(2371) INFO: <core> [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(2371) INFO: cdp [mod.c:227]: cdp_child_init(): CDiameterPeer child starting ... 0(2371) INFO: cdp [mod.c:229]: cdp_child_init(): ... CDiameterPeer child started 3(2381) INFO: cdp [worker.c:331]: worker_process(): [0] Worker process started... 4(2382) INFO: cdp [worker.c:331]: worker_process(): [1] Worker process started... 5(2383) INFO: cdp [worker.c:331]: worker_process(): [2] Worker process started... 7(2390) INFO: cdp [receiver.c:445]: receiver_process(): receiver_process(): [] Receiver process doing init on new process... 7(2390) INFO: cdp [receiver.c:450]: receiver_process(): receiver_process(): [] Receiver process starting up... 8(2392) INFO: cdp [receiver.c:445]: receiver_process(): receiver_process(): [hss.net1.test] Receiver process doing init on new process... 8(2392) INFO: cdp [receiver.c:183]: add_serviced_peer(): add_serviced_peer(): Adding serviced_peer_t to receiver for peer [hss.net1.test] 8(2392) INFO: cdp [receiver.c:450]: receiver_process(): receiver_process(): [hss.net1.test] Receiver process starting up... 9(2393) INFO: cdp [acceptor.c:82]: acceptor_process(): Acceptor process starting up... 9(2393) WARNING: cdp [tcp_accept.c:123]: create_socket(): create_socket(): Trying to open/bind/listen on 10.0.0.9 port 3870 9(2393) WARNING: cdp [tcp_accept.c:148]: create_socket(): create_socket(): Successful socket open/bind/listen on 10.0.0.9 port 3870 9(2393) INFO: cdp [acceptor.c:96]: acceptor_process(): Acceptor opened sockets. Entering accept loop ... 10(2394) INFO: cdp [timer.c:205]: timer_process(): Timer process starting up... 10(2394) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 0 10(2394) INFO: cdp [peerstatemachine.c:526]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer hss.net1.test 10(2394) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 10.0.0.9 port 3868 10(2394) WARNING: cdp [receiver.c:911]: peer_connect(): peer_connect(): Error opening connection to to 10.0.0.9 port 3868 >Connection refused 10(2394) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 0 10(2394) INFO: cdp [peerstatemachine.c:526]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer hss.net1.test 10(2394) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 10.0.0.9 port 3868 10(2394) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.net1.test:3868 connected 10(2394) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 6 10(2394) WARNING: cdp [peermanager.c:315]: peer_timer(): Inactivity on peer [hss.net1.test], sending DWR... - if we don't get a reply, the peer will be closed 10(2394) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 6 8(2392) INFO: cdp [receiver.c:802]: receive_loop(): select_recv(): [hss.net1.test] read on socket [15] returned 0 > ... dropping 8(2392) INFO: cdp [receiver.c:219]: disconnect_serviced_peer(): drop_serviced_peer(): [hss.net1.test] Disconnecting from peer 10(2394) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 0 10(2394) INFO: cdp [peerstatemachine.c:526]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer hss.net1.test 10(2394) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 10.0.0.9 port 3868 10(2394) WARNING: cdp [receiver.c:911]: peer_connect(): peer_connect(): Error opening connection to to 10.0.0.9 port 3868 >Connection refused ^C11(2398) INFO: <core> [main.c:798]: sig_usr(): signal 2 received 9(2393) INFO: <core> [main.c:798]: sig_usr(): signal 2 received 0(2371) ALERT: <core> [main.c:725]: handle_sigs(): child process 2381 exited normally, status=0 0(2371) INFO: <core> [main.c:740]: handle_sigs(): dont_fork turned on, living on 0(2371) NOTICE: <core> [main.c:683]: handle_sigs(): Thank you for flying kamailio!!! 10(2394) INFO: <core> [main.c:798]: sig_usr(): signal 15 received 0(2371) INFO: cdp [mod.c:242]: cdp_exit(): CDiameterPeer child stopping ... On Fri, Jan 22, 2016 at 2:16 AM, Franz Edler <franz.edler(a)technikum-wien.at<mailto:franz.edler@technikum-wien.at>> wrote: Hi, you can see the first error is a database error: ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on 'server.net1.test' (111) This error I expected because you have to adapt the access permissions to the database. Just check the GRANT statements in my log-file and try to understand what they do. Then adapt it accordingly. BR Franz From: Ramya Y [mailto:y.ramya146@gmail.com<mailto:y.ramya146@gmail.com>] Sent: Wednesday, January 20, 2016 2:19 PM To: franz.edler(a)technikum-wien.at<mailto:franz.edler@technikum-wien.at>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Subject: Re: [SR-Users] Kamailio IMS deployment Hi Franz we have tried with different interfaces(eth0(10.0.0.9),eth1(10.0.0.10),eth2(10.0.0.11),eth3(10.0.0.12) for P/I/S ,but facing below error. root@server:/etc# kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(1630) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: pcscf.net1.test [10.0.0.10]:4060 Aliases: *: pcscf.net1.test:* WARNING: no fork mode 0(1630) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(1630) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(1630) INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 0(1630) ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on 'server.net1.test' (111) 0(1630) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(1630) ERROR: ims_usrloc_pcscf [usrloc_db.c:79]: init_db(): unable to connect to the database 0(1630) ERROR: ims_usrloc_pcscf [ul_mod.c:203]: mod_init(): Error initializing db connection 0(1630) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module ims_usrloc_pcscf (/usr/local/lib64/kamailio/modules/ims_usrloc_pcscf.so) ERROR: error while initializing modules 0(1630) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialize could you please help me how to proceed further, steps followed: =========== 1)Modified at - /etc/network/interfaces: created eth1,eth2,eth3 for P/I/S-CSCF. Interfaces are up. 2)Added Bind address at /etc/mysql/my.cnf bind-address 10.0.0.9 bind-address 10.0.0.10 bind-address 10.0.0.11 bind-address 10.0.0.12 3)Modified at /etc/bind/net1.test.zone $ORIGIN net1.test. $TTL 1W @ 1D IN SOA serverkde. root.net1.test ( 2010081401 ; Serial YYYYMMDDXX 3H ; refresh 15M ; retry 1W ; expiry 1D ) ; minimum 1D IN NS server server 1D IN A 10.0.0.9 hss 1D IN CNAME server as1 1D IN CNAME server as2 1D IN CNAME server xcap 1D IN CNAME server server1 1D IN A 10.0.0.10 pcscf 1D IN CNAME server1 server2 1D IN A 10.0.0.11 icscf 1D IN CNAME server2 server3 1D IN A 10.0.0.12 scscf 1D IN CNAME server3 4)Modified at /etc/bind/0.0.10.zone $TTL 86400 @ IN SOA server. root.net2.test ( 2010081401 ; Serial YYYYMMDDXX 28800 ; Refresh 14400 ; Retry 3600000 ; Expire 86400 ) ; Minimum IN NS server.net1.test. IN NS server1.net1.test. IN NS server2.net1.test. IN NS server3.net1.test. 10 IN PTR notebook.net1.test. 9 IN PTR server.net1.test. 7 IN PTR server.net2.test. 5)Modified at /etc/resolve.conf domain net1.test search net1.test nameserver server.net1.test nameserver server1.net1.test nameserver server2.net1.test nameserver server3.net1.test 6)Modified at /etc/hosts 127.0.0.1 localhost 10.0.0.9 server.net1.test server 10.0.0.10 server1.net1.test server1 10.0.0.11 server2.net1.test server2 10.0.0.12 server3.net1.test server3 10.0.0.9 net1.test 10.0.0.10 pcscf.net1.test 10.0.0.11 icscf.net1.test 10.0.0.12 scscf.net1.test Thanks, Ramyasri. On Sun, Jan 17, 2016 at 5:33 PM, Franz Edler <franz.edler(a)technikum-wien.at<mailto:franz.edler@technikum-wien.at>> wrote: Fine. This exactly was my intention. BR Franz > -----Original Message----- > From: sr-users [mailto:sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>] On Behalf Of > Vasiliy Ganchev > Sent: Monday, January 11, 2016 9:08 AM > To: sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio IMS deployment > > Hi Franz! > > Thanks for sharing! Great job, end excellent materials to get started with > students (beside my main work I also teach students in Odessa National > Academy of Telecommunication). > I was looking for some "light" way how to start with real deployment of IMS, > not pure theory, and this is just the kind I wanted! > > Will try your deploy in education process this year. (will share the results in > the end) > > Cheers! > > > > -- > View this message in context: http://sip- > router.1086192.n5.nabble.com/Kamailio-IMS-deployment-<http://router.1086192.n5.nabble.com/Kamailio-IMS-deployment-> > tp143593p144517.html > Sent from the Users mailing list archive at Nabble.com. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr- > users(a)lists.sip-router.org<mailto:users@lists.sip-router.org> http://lists.sip-router.org/cgi- > bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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A new edition of Kamailio Development Workshop
by Daniel-Constantin Mierla 27 Jan '16

27 Jan '16

Hello, last edition of Kamailio Development Workshop happened about three years ago. As I got requests about it from time to time, I thought of trying to organize a new edition. Rather short term by now due to various constraints, so the plan is to do it again in Alicante, Spain, during February 15-16, 2016. I put together more details at: * http://www.kamailio.org/w/development-workshop/ Besides spending 2 days digging into Kamailio C code, such event is also good for discovering and sharing development resources among participants, as well as meeting people from community -- at the previous editions I met for first time with Victor Seva, Vicente Hernando and Seudin Kasumovic, which in turn contributed a lot of code afterwards. It will be also testing phase for upcoming major release, maybe we can plan some stress testing session on site. Right now I am looking to see if there is any interest in such event in order to nail down organizing it or not. Should anyone consider to participate, write me. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

2 4

Acc message to syslog
by Yasin CANER 27 Jan '16

27 Jan '16

4 3

Strange PUA Behaviour
by Phil Lavin 26 Jan '16

26 Jan '16

Hi all, We are investigating an issue with BLF whereby the BLF key on the subscribed phone does not always "turn off" after the call has ended. We have enabled force_single_dialog to keep things simple for debugging. The attached pcap is the result. The relevant stuff as I see it is: * Packet #60 - UA sends a BYE * Packet #62 - PUA sends a PUBLISH over loopback with a state of "terminated" * Packet #65 - The PUBLISH is sent out to the subscribed UA as a NOTIFY with a state of "early" This causes the BLF light on the phone to continue flashing. We are running a version of Kamailio compiled from the HEAD of the current 4.3 branch. Any ideas? Phil Lavin Telecoms Systems Manager CloudCall by SYNETY www.cloudcall.com<http://www.cloudcall.com/> T: +44 (0) 330 335 0000 / +1 617 982 1600 D: +44 (0) 116 424 4790 / +1 617 982 4790 SM: LinkedIn<https://uk.linkedin.com/pub/phil-lavin/25/422/750> READ OUR BLOG FOR SMARTER COMMUNICATIONS<http://t.sidekickopen03.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XX43…> Confidentiality: This e-mail transmission, including any attachments, is intended only for the named recipient(s) and may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. If you have received this transmission in error, or are not the named recipient(s), please notify the sender immediately by return e-mail and permanently delete this transmission, including any attachments. Security: This e-mail and any attachments are believed to be free from any virus but it is the responsibility of the recipient to ensure this is so. E-mail is not a 100% secure communication.

2 19

IPTel URI forwarding to PBXes not working
by McConnachie, Alex 26 Jan '16

26 Jan '16

I am stumped as my DID provider can't route via SIP URI to PBXes but they can to IPTel. I can call my own PBX, and from my own PBX to the PBXes URI without issue. When I route the DID via URI to IPTel and set voicebox it works and routes to the voicebox. When I add a forward for the PBXes URI it then breaks and gets a denial of 404 not in service. Ideas anyone? I am scratching my head here. Appreciate any help! Regards, Alex McConnachie

2 1

Rewriting header of BYE message to carrier
by Ryan Mottley 26 Jan '16

26 Jan '16

Hi, We're running a system with Kamailio running in front of Asterisk just handling registrations and forwarding everything else to Asterisk. But we're having an issue during hangup on incoming calls. If the initiator hangs up, the call completes successfully. But if one of our phones hangs up, the BYE message comes back with a 404 "Not Found" and the call doesn't hang up on the carrier side. According to the carrier, it's because the IP in the contact on our ACK message goes to their audio IP while the header of our BYE points to their signaling IP. ACK sip:[Kamailio Pub IP]:5060;line=sr--rkpsDAp6YAp6DIpZDZmZeI2ZYI26YIRVDcpsDIpsem* SIP/2.0 Via: SIP/2.0/UDP *[Carrier Signaling IP]*;branch=z9hG4bK2236.1402e7b4.2 Via: SIP/2.0/UDP *[Carrier Audio IP]*;received=*[Carrier Audio IP]* ;branch=z9hG4bK07a8bccb;rport=5060 Route: <sip:[Kamailio Pub IP];r2=on;lr=on;ftag=as67cef00d;nat=yes>,<sip:10.120.0.1;line=sr--rkpsDthVDIhVDNh6Ogo6eKh6eAQs4LRflC2srQRflC2srGqAl-CAP6rZrZkGDmpGed2APtCvlx1> From: "+16014477389" <sip:6014477389@*[Carrier Audio IP]*>;tag=as67cef00d To: <sip:6016025063@*[Carrier Signaling IP]*>;tag=as643b40ca Contact: <sip:6014477389@*[Carrier Audio IP]*> Call-ID: 4aaefec90826a2a221f0af9500ad211b@*[Carrier Audio IP]* CSeq: 102 ACK User-Agent: packetrino Max-Forwards: 69 Content-Length: 0 BYE sip:6014477389@*[Carrier Signaling IP] *SIP/2.0 Via: SIP/2.0/UDP [Kamailio Pub IP];branch=z9hG4bK2236.fca983a45913fb510f97e781a85c7392.0 Via: SIP/2.0/UDP 10.120.0.1;branch=z9hG4bKsr-IqktV1L26BCx0jmwZeI2ZYI26YIRVDcpsDIpsem.-EF8-EtCZYmg6edIMehhA4A.AEzyuiZKfPKo7N-qAcWq6D-rsYc4Zp** Route: <sip:*[Carrier Signaling IP]*;lr=on> Max-Forwards: 69 From: <sip:6016025063@[Kamailio Pub IP]>;tag=as643b40ca To: "+16014477389" <sip:6014477389@*[Carrier Audio IP]*>;tag=as67cef00d Call-ID: 4aaefec90826a2a221f0af9500ad211b@*[Carrier Audio IP]* CSeq: 102 BYE User-Agent: Asterisk PBX 13.6.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 I'm thinking it's happening because their side isn't configured correctly to handle traffic coming back from a proxy, but in the meantime is there a way to rewrite the top of the BYE header to match the "audio IP" they're requesting it be sent to? Thanks! -- Ryan Mottley, Developer VOXO, LLC voxo.co - (601)602-5063

2 1

Missing dependencies for kamailio-mysql-4.2.7-3.1.x86_64
by Zuxy Meng 26 Jan '16

26 Jan '16

Hi there, I'm trying to install this rpm for my RHEL 5 box, but it asks for mysql-libs; but RHEL 5 isn't shipped with mysql-lib, only mysql55-mysql-libs, which kamailio-mysql doesn't recognize as a valid dependency. Any workaround here? Thanks! -- Zuxy Beauty is truth, While truth is beauty. PGP KeyID: E8555ED6

2 5

Fwd: [Free-RTC] FOSDEM Real Time devoom: call for help
by Daniel Pocock 26 Jan '16

26 Jan '16

-------- Forwarded Message -------- Subject: [Free-RTC] FOSDEM Real Time devoom: call for help Date: Sun, 24 Jan 2016 19:17:02 +0100 From: Saúl Ibarra Corretgé <saul(a)ag-projects.com> Reply-To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org> To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org>, summit(a)xmpp.org Hi all, FOSDEM is less than a week away, I hope you are all as excited about it as I am :-) We need some volunteers to help with the video recording gear on Saturday. You danât have to do it for the entire day, but the more volunteers we have the better the recording will be.. hopefully! NB: volunteers get a reserved seat, so in case the room is full youâd still be able to enjoy a seat. Please feel free to forward this message to anyone who might be interested. Thanks! -- SaÃºl Ibarra CorretgÃ© AG Projects

1 0

question about kamailio handling of forwarded invite
by DING MA 26 Jan '16

26 Jan '16

Hi, all We're trying to build a system that consists of pbx, kamailio and asterisk in the following configuration. pbx (sip trunk) --- kamailio --- asterisk The kamailio and asterisk are integrated with same database. The outgoing calls to pbx works. But there is a problem with incoming calls from pbx. If we make a consecutive calls from the same pbx user to the same user registered with kamailio. The first would go through, but the second call would be rejected by asterisk. We have insecure=invite set on the trunk/peer, so asterisk is not supposed to auth the invite from kamailio. But the pbx user (from in this case) is not in the database. The asterisk log says: [Jan 21 23:13:19] VERBOSE[20785] chan_sip.c: --- (16 headers 13 lines) --- [Jan 21 23:13:19] VERBOSE[20785] chan_sip.c: Sending to 10.0.1.30:5061 (no NAT) [Jan 21 23:13:19] VERBOSE[20785][C-00000001] chan_sip.c: Sending to 10.0.1.30:5061 (no NAT) [Jan 21 23:13:19] VERBOSE[20785][C-00000001] chan_sip.c: Using INVITE request as basis request - 4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061 [Jan 21 23:13:20] NOTICE[20785][C-00000001] acl.c: SIP Peer ACL: Rejecting '10.0.1.30' due to a failure to pass ACL '(BASELINE)' [Jan 21 23:13:20] NOTICE[20785][C-00000001] chan_sip.c: Failed to authenticate device <sip:95678@10.0.1.35>;tag=as4028dabf [Jan 21 23:13:20] VERBOSE[20785][C-00000001] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.0.1.30:5061 ---> SIP/2.0 403 Forbidden^M Via: SIP/2.0/TLS 10.0.1.30:5061 ;branch=z9hG4bK9c8e.5cd2c05f6a572312c7793abf5fe1183c.0;i=2;received=10.0.1.30^M Via: SIP/2.0/TLS 10.0.1.35:5061 ;received=10.0.1.35;branch=z9hG4bK249855c1;rport=59929^M From: <sip:95678@10.0.1.35>;tag=as4028dabf^M To: <sip:16317@10.0.1.30>;tag=as35f47241^M Call-ID: 4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061^M CSeq: 102 INVITE^M Server: Asterisk PBX 13.6.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE^M Supported: replaces, timer^M Content-Length: 0^M Comparing the two invites from kamailio to asterisk, it seems the only difference is that the second invite has an "i=2" in the Via header while the first one has "i=1". Not sure what the "i=1" is for. Would appreciate some insights on how kamailio is adding/handling the "i=#" in Via header. Thanks. Ding Ma SPG, Motorola Solutions

3 2

Acc leaking
by Yasin CANER 25 Jan '16

25 Jan '16

3 3

kamailio_call_disconnecting issue
by yasir ay 25 Jan '16

25 Jan '16

Hello, Hi team I have a kamailio SIP server for call Routing,my call flow is like VoipDialer--->Kamailio-Routing Server-->Softswitch-->Telco/PSTN Mobile caller----------------------------------------------------------------------->calle Now Am facing an issue for a call from voip to telco after answering the call i tried to disconnect.but call is not disconneting..what will i do in kamailio.cfg pls help me to re-write the script am using tls also Regards AmanIdo

2 1

using myself in equation with from_uri
by Dirk Teurlings - SIGNET B.V. 25 Jan '16

25 Jan '16

Hi, The documentation states the following. https://www.kamailio.org/wiki/cookbooks/4.2.x/core#myself if(uri==myself) { log("the request is for local processing\n"); }; Now this could also be used to do something like this: (also used in some examples) if(from_uri==myself) { log("the request is for local processing\n"); }; Using this the from_uri could be: sip:sip.voipexample.net:5061 myself in this case only has sip.voipexample.net:5060 as an Aias Now the equation wouldn't match, because the port 5061 is non-default. Would it be possible to match the from_uri to myself ignoring the port in from_uri? The only thing I can think of is using a regular expression. somthing like this: from_uri=~"sip:.+@sip\.voipexample\.net(:([0-9]{1,4}|[1-5][0-9]{4}|6[0-4][0-9]{3}|65[0-4][0-9]{2}|655[0-2][0-9]|6553[0-5])|)" But this skips the use of myself entirely. Any insight on this would be greatly appreciated! Cheers, Dirk

3 7

poros@@lists.sip-router.org
by Poros Mitra 25 Jan '16

25 Jan '16

1 0

Fosdem 2016
by Daniel-Constantin Mierla 22 Jan '16

22 Jan '16

Hello, anyone going to Fosdem? If no unexpected weather conditions messing the transport system, I am going to be there. It would be good to know who else we can meet there. If there are enough people interested, we can try organizing a social event (dinner/drinks) on Saturday evening. Last year we did it together with Jitsi team, but they are not coming this year. Perhaps there are other folks around the RTC dev room we can ask. On the other hand, it is harder to find a place if it is a large group, even harder to put all together if there are different groups/coordinators. Also, it is not easy to discuss withing a large group, as typically for a dinner we were seated to tables. Therefore organizing for a smaller group just for Kamailio friends could be a good alternative. Let's see first the number of participant and the level of interest for various options. This being an announcement, was crossposted to few mailing lists, but as most of the people are on sr-users, I think it would be good to reply and follow up on this discussion only there. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

5 5

Kamailio and openLDAP integration.
by Safdar Khan 22 Jan '16

22 Jan '16

Hi guys, I have tried to integrate kamailio and ldap.It seems like both are taking to each other. Here the purpose of ldap is for back-end support for sip user authentication rather than kamailio database.However i am able to log in but could not make call between sip users.I could not figure out what i am missing. Here is the routing block for ldap in my kamailio.cfg > request_route { > route(LDAPAUTH); > > } --- route[LDAPAUTH] > { > if(is_method("REGISTER")) > { > if(is_present_hf("Authorization")) > { > if > (!ldap_search("ldap://sipaccounts/ou=MyUsers,dc=example,dc=com?cn,userPassword?one?(cn=$fU)")) > { > switch ($retcode) > { > case -1: > sl_send_reply("404", "User Not Found"); > exit; > case -2: > sl_send_reply("500", "Internal server error"); > exit; > default: > exit; > } > } > ldap_result("cn/$avp(username)"); > ldap_result("userPassword/$avp(password)"); > route(REGISTRAR); > if (!pv_www_authenticate("$td", "$avp(password)", "4")) > { > www_challenge("$td", "1"); > exit; > } > sl_send_reply("200", "ok"); > exit; > } > else > { > www_challenge("$td", "1"); > exit; > } > } > }

3 3

SIP Trunking using Kamailio IMS
by indranil chowdhury 21 Jan '16

21 Jan '16

Hello, Is SIP trunking possible with the free version of Kamailio or we need to upgrade to business license? Kindly get back to us with the details of using SIP trunking using Kamailio. We have a requirement in our organisation regarding the same which uses IMS and SIP trunking. In this setup we would initially like to test with 2 LTE EPCs connected to two Kamailio IMS with a SIP trunk between them. Thanks and Regards, Indranil

3 2

topoh quirk with 200 OK for CANCEL
by Alex Balashov 21 Jan '16

21 Jan '16

Hi, I'm using topoh and am running into lots of these errors: Jan 19 20:34:52 csrpus1 /usr/local/sbin/kamailio[3641]: ERROR: [R-DEFAULT-STATELESS-REPLY:nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc] !> Received stateless reply 200 (CANCEL) from xxx.xxx.xxx.xxx:5060 Jan 19 20:34:53 csrpus1 /usr/local/sbin/kamailio[3639]: ERROR: tm [t_lookup.c:932]: t_reply_matching(): matching transaction found but callids don't match (received: nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc stored: 51899BC6-569EE4330005CE1B-7D47C700_bleg) The call topology is: UAC --> Kamailio (+ topoh) --> PSTN GW Topoh is behaving precisely as expected, but it looks like we might have hit a TM case that topoh does not accommodate somehow: 1. A CANCEL comes in from the UAC and Kamailio sends a CANCEL for its branch to the PSTN GW. 2. Kamailio sends a '200 cancelling' for the UAC's branch backward. 3. The PSTN GW replies with a 200 OK for the CANCEL (PSTN GW branch) that contains the topoh-spoofed Call-ID, but for some reason Kamailio does not match this reply, as indicated by the above error. 4. Because the 200 OK reply was stateless and not matched to a CANCEL transaction, this causes Kamailio to retransmit the CANCEL to the PSTN GW, at which point it receives: 481 Call Leg/Transaction Does Not Exist 5. #4 repeats once more. I don't see any real harm from this situation, but it seems like something that needs fixing. The 487 Request Terminated final reply is processed and relayed correctly to the UAC, and the dialog ends appropriately everywhere. FWIW, I thought this might have to do with async processing of the initial INVITE (through mqueue + rtimer), so I turned it off, but it didn't make any difference. My CANCEL handling in the main request route looks like this: if(is_method("CANCEL")) { set_rtpengine_set("1"); rtpengine_delete(); if(!t_relay_cancel()) { sl_send_reply("500", "Internal Server Error"); exit; } xlog("L_ERR", "[R-MAIN:$ci] !> " "Corresponding INVITE transaction for CANCEL " "does not (or, no longer) exists\n"); exit; } So, it just looks like TM is not matching, and therefore not absorbing the 200 OK reply to its CANCEL to the upstream gateway -- when topoh is enabled. Any insights would be appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 11

RFC: removing (disabling) fork=no
by Daniel-Constantin Mierla 21 Jan '16

21 Jan '16

Hello, wondering if anyone is using fork=no -- some old docs suggest it is suitable for debugging, but actually kamailio doesn't work properly in this mode, leading to more troubles than benefits (e.g., having reports of invalid issues, like tcp not working in this mode). In first phase I would disable setting this value, with a warning if set to no, because most of the configs out there have fork=yes. Removing it could be considered in the future. Note that this fork=no is different than don't daemonize controlled with -D, which will stay being useful for some init.d systems. Comments or other suggestions? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

2 2

Re: [SR-Users] Fwd: Kamailio and NAT
by Nelson Migliaro 20 Jan '16

20 Jan '16

I am thinking that I cat get the port during the first call and then use that port for the rest of calls. Maybe that first call will fail but after that, all calls will go fine. Example of a Trying showing received port: 52548 ¿Do you know what pseudovariable represent that value? I tested with all variables I could find but I could not find anyone. Thank you --------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/01/19 17:38:38.981987 VENDOR-IP:5060 -> KAMAILIO-IP:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP PUBLIC-IP:52548;branch=z9hG4bK8b5c.b122371ac0ac2f3ef3204b0f192cb16c.1;rport=52548 Via: SIP/2.0/UDP ASTERISK-IP:5060;received=ASTERISK-IP;branch=z9hG4bK4082d124;rport=5060 From: 888888888 <sip:888888888@ASTERISK-IP>;tag=as34f971fe To: <sip:999999999@VENDOR-DOMAIN> Call-ID: 0b5307fa290674a97b47970643fce42a@ASTERISK-IP:5060 CSeq: 102 INVITE Server: kamailio Content-Length: 0 2016-01-15 9:50 GMT+01:00 Nelson Migliaro <eng.migliaro(a)gmail.com>: > Hello Daniel, > > Yes, I am registered to the vendor. > > Regards, > > Nelson.- > > 2016-01-15 7:58 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Ahh, I thought Asterisk is in the public internet, but actually you >> connect to a provider (vendor), which seems to run Kamailio as well. >> >> Using information from 100 trying is too late, as the INVITE was already >> sent... so one more question before trying to propose a solution. Do you >> have to register to the provider? >> >> Cheers, >> Daniel >> >> >> On 14/01/16 18:51, Nelson Migliaro wrote: >> >> Yes it is possible, but is there an easy way to workaround the issue >> using Kamailio. >> >> Because I have the port because vendor is sending that info in Trying: >> >> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >> SIP/2.0 100 trying -- your call is important to us >> Via: SIP/2.0/UDP PUBLIC-IP:52548;branch= >> z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=*52548* >> >> I need to find the way to discover the port used by firewall (maybe >> getting that info from Trying) and then advertise that port. >> >> >> >> 2016-01-14 18:32 GMT+01:00 Daniel-Constantin Mierla < <miconda(a)gmail.com> >> miconda(a)gmail.com>: >> >>> Not really up to date with all Asterisk features -- do you know if you >>> can append a custom header to a SIP response that is going to be generated >>> by Asterisk? Eventually the reply for an OPTIONS request. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 14/01/16 17:19, Nelson Migliaro wrote: >>> >>> Yes, I manage all devices, even the internet router but it does not >>> allow static pat. >>> >>> 2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>> >>>> Do you control the Asterisk? If yes, depending on Asterisk capabilities >>>> of building replies, you may be able to do some automation to detect the >>>> external port. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro < >>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote: >>>> >>>>> There is not a public Kamailio, only one Kamailio behind NAT, >>>>> >>>>> Right now the configuration is: >>>>> >>>>> Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy ) >>>>> <-> Internet router (public IP + symmetric na) <-> Internet >>>>> >>>>> Regards, >>>>> >>>>> 2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla < >>>>> <miconda(a)gmail.com>miconda(a)gmail.com>: >>>>> >>>>>> Is the kamailio behind nat communicating with another kamailio on a >>>>>> public IP? >>>>>> >>>>>> Cheers, >>>>>> DAniel >>>>>> >>>>>> On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro < >>>>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote: >>>>>> >>>>>>> Thank you Daniel for your answer, >>>>>>> >>>>>>> As you mention, there is a symmetric nat and router does not allow a >>>>>>> static NAT. >>>>>>> >>>>>>> By sniffing traffic I can see the port is using new but in case it >>>>>>> change, how can automate the process of advertising the correct port? >>>>>>> >>>>>>> Cheers! >>>>>>> >>>>>>> >>>>>>> ---------- Forwarded message ---------- >>>>>>> From: Daniel-Constantin Mierla < <miconda(a)gmail.com> >>>>>>> miconda(a)gmail.com> >>>>>>> Date: 2016-01-13 23:28 GMT+01:00 >>>>>>> Subject: Re: [SR-Users] Kamailio and NAT >>>>>>> To: "Kamailio (SER) - Users Mailing List" < >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org> >>>>>>> >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> it looks like you have a symmetric nat router, so the allocated port >>>>>>> is randomly selected. >>>>>>> >>>>>>> If you don't control the nat router to set a static forwarding rule >>>>>>> or it doesn't provide the option to set static forwarding, then you are >>>>>>> pretty much left with sniffing the traffic to discover the external port >>>>>>> and advertise it. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 13/01/16 20:31, Nelson Migliaro wrote: >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I finally were able to run my Kamailio behind NAT but in order to >>>>>>> accomplish that I included: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 >>>>>>> >>>>>>> 52548 is the port my internet router change when doing NAT >>>>>>> (5060->52548). I found this port sniffing traffic >>>>>>> >>>>>>> Conclusions at this point are: >>>>>>> >>>>>>> >>>>>>> ---------------------------------------------1-------------------------------------------------------------------------------------------------- >>>>>>> If I use this line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work >>>>>>> :( >>>>>>> >>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because >>>>>>> they are part of the same transaction >>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port >>>>>>> 5060 and router blocks de request. I assume vendor sends BYE to 5060 >>>>>>> because it is a new transaction >>>>>>> >>>>>>> -----------------------------------------------2-------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> If I use this line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!! >>>>>>> >>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because >>>>>>> they are part of the same transaction >>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port >>>>>>> 52548 and router forward the request to Kamailio. Since there is an open >>>>>>> connection. >>>>>>> >>>>>>> I need to find the way to find the way to advertise the public port >>>>>>> internet router is doing NAT (PAT). >>>>>>> >>>>>>> >>>>>>> --------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> This trace is a call that worked fine because I included line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 >>>>>>> >>>>>>> >>>>>>> This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060 >>>>>>> advertise PUBLIC-IP:52548 >>>>>>> 2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060 >>>>>>> INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0 >>>>>>> Record-Route: < >>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB >>>>>>> A-;nat=yes> >>>>>>> Via: SIP/2.0/UDP >>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0 >>>>>>> Via: SIP/2.0/UDP >>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060 >>>>>>> Max-Forwards: 69 >>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453 >>>>>>> To: <sip:NUM-DESTINATION@sip.VENDOR-IP> >>>>>>> Contact: >>>>>>> <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1> >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 102 INVITE >>>>>>> User-Agent: Kamailio >>>>>>> Date: Wed, 13 Jan 2016 19:10:15 GMT >>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >>>>>>> INFO, PUBLISH, MESSAGE >>>>>>> Supported: replaces, timer >>>>>>> Content-Type: application/sdp >>>>>>> Content-Length: 255 >>>>>>> >>>>>>> >>>>>>> Trying..... >>>>>>> >>>>>>> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >>>>>>> SIP/2.0 100 trying -- your call is important to us >>>>>>> Via: SIP/2.0/UDP >>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548 >>>>>>> Via: SIP/2.0/UDP >>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060 >>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453 >>>>>>> To: <sip:NUM-DESTINATION@VENDOR-IP> >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 102 INVITE >>>>>>> Server: kamailio >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> And finally a BYE >>>>>>> >>>>>>> 2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >>>>>>> BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1 >>>>>>> SIP/2.0 >>>>>>> Via: SIP/2.0/UDP >>>>>>> VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0 >>>>>>> From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93 >>>>>>> To: "NUM-SOURCE" <sip:NUM-SOURCE@VENDOR-IP>;tag=as3b72a453 >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 28731 BYE >>>>>>> Max-Forwards: 69 >>>>>>> Route: < >>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na >>>>>>> yes> >>>>>>> Reason: Q.850;cause=16 >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> Finally, It is finally working because I hardcoded NAT´d port. >>>>>>> I would like to find a way to avoid setting the port in "hard". >>>>>>> >>>>>>> Thank you >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>> list >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org >>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>> list >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org >>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla - <http://www.asipto.com> >>>>>> http://www.asipto.com >>>>>> <http://twitter.com/#%21/miconda>http://twitter.com/#!/miconda - >>>>>> <http://www.linkedin.com/in/miconda>http://www.linkedin.com/in/micond >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>> list >>>>>> sr-users(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users(a)lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - <http://www.asipto.com>http://www.asipto.com >>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond >>>> <http://www.linkedin.com/in/miconda> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >> >> >

2 2

Registration to sip provider
by Evren Akbulut 19 Jan '16

19 Jan '16

Hi, I am a newbie. As I read I understand I need to use UAC module for registration but cannot figure how to trigger the registration request. I enable the UAC module and it seem to be working. How can I trigger the REGISTER to a SIP provider account? And is it possible to use multiple accounts for different provider registrations? Can someone help about UAC usage please. Regards Evren Akbulut

2 1

rtpengine to do DTLS-SRTP handshake, but media should not pass through it. the media should go through media server, which is running on the same machine
by riko nir 19 Jan '16

19 Jan '16

Hi, We have a media server and (opensips + rtpengine). How can I configure opensips+rtpengine to just do DTLS handshake with the client, but the media should not go through RTPEngine. The media should go to the Media server. Thanks. Riko

1 0

Kamailio Load Balancing Cluster
by sandeep goje 19 Jan '16

19 Jan '16

Hi, I am using Kamailio as a load balancer with call load distribution to Freeswitch Gateways. I need a HA cluster of Kamailio. I have seen instructions to setup Kamailio HA cluster using KeepAlive but I also want them to share the load balancing information. I have seen OpenSIPS supports this. Is there a way in Kamailio to make both load balancers also share the load balancing information. Thank you. Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston

2 1

Get value of RPID privacy param
by Grant Bagdasarian 18 Jan '16

18 Jan '16

Hello, How do I select the value of the privacy parameter in the Remote-Party-ID header? Remote-Party-ID: "John Doe" <sip:jdoe@foo.com>;party=calling;id;privacy=full;screen=yes The following doesn't work, and I believe because it's not a valid select class: $sel((a)hf_value.remote_party_id.privacy) I'm using kamailio 4.1. Regards, Grant

5 9

IMS terminating dialog does not re-run through all iFCs after calling t_newtran()
by Russell McConnachie 17 Jan '16

17 Jan '16

Hi, My CSCF environment has an application server acting as a B2BUA. It's originating a new session, and creating a new terminating session back to the S-CSCF (two separate sessions per-call). When I perform a re-INVITE with a diversion to transfer the call to another URI it is all good until the S-CSCF attempts to call the call control logic: # Start new transaction: t_newtran(); if (isc_match_filter("orig", "location")) { t_on_failure("ISC_ORIG_FAILURE"); exit; The t_newtran() method from what I understand will create a new transaction. Inside of this new transaction, I would think that the isc_match_filter should re-run through all of the iFCs defined on the HSS service profile, but it does not seem to be doing so. It seems to be continuing at where the original transaction ISC mark skip point was set. Thanks

2 1

Different rtpproxy for different media type
by Koray Vatansever 15 Jan '16

15 Jan '16

Hi everybody, Is there a way to use different rtpproxies for different media types? I want to use one rtpproxy set for audio, and use another one for video. Is this possible in kamailio? Regards, Koray

2 5

Kamailio replies with SIP/2.0 500 Service Unavailable Under load
by sandeep goje 15 Jan '16

15 Jan '16

Hi, I am doing load testing with of Kamailio. I have sipp (version 3.3) as the load generator,Kamailio(4.2.7) as Load Balancer and Freeswitch (1.6.5) as Gateways. I am using the following command to generate load ./sipp -sf uac.xml -d 2000 -m 4000 -l 1000 -r 200 -trace_stat -i SIPP_IP -p 15060 --trace_err -trace_error_codes -trace_calldebug -trace_screen KAM_IP When the rate of generation is below 80 (-r 80), Kamailio behaves fine.But if we go beyond 80, Kamailio starts sending "500 Service Unavailable" 2016-01-12 11:18:15.375103 1452577695.375103: Aborting call on unexpected message for Call-Id '96-32493@SIPP_IP': while expecting '180' (index 2), received 'SIP/2.0 500 Service Unavailable^M Via: SIP/2.0/UDP SIPP_IP:15060;branch=z9hG4bK-32493-96-0^M From: sipp <sip:sipp@SIPP_IP:15060>;tag=96^M To: sut <sip:service@KAM_IP :5060>;tag=55f576f507e822fa6633cf4bc22740e6-6660^M Call-ID: 96-32493@SIPP_IP^M CSeq: 1 INVITE^M Server: kamailio (4.2.7 (x86_64/linux))^M Content-Length: 0^M With the above command, sipp output is ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2016-01-12 11:26:07.723764 1452578167.723764 Last Reset Time | 2016-01-12 11:27:16.280901 1452578236.280901 Current Time | 2016-01-12 11:27:16.282737 1452578236.282737 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:001000 | 00:01:08:558000 Call Rate | 0.000 cps | 58.345 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 4000 Total Call created | | 4000 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 1192 Failed call | 0 | 2808 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000000 | 00:00:15:102000 Call Length | 00:00:00:000000 | 00:00:10:614000 ------------------------------ Test Terminated -------------------------------- Kamailio is run with command /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 8 -u kamailio -g kamailio version: kamailio 4.2.7 (x86_64/linux) 727746 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 727746 compiled on 15:59:03 Jan 11 2016 with gcc 4.6.3 System : Ubuntu 12.04.4 LTS Linux devops10-60-20-169 3.11.0-15-generic #25~precise1-Ubuntu SMP Thu Jan 30 17:39:31 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux I am not seeing any errors in Kamailio Logs. Any pointers on how to get it to serve more calls/sec Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston

4 3

Re: [SR-Users] Parser error when enabling siptrace with HEP functionality
by Sergey Basov 15 Jan '16

15 Jan '16

And one more thing. In captured packet from Homer, I see original SIP packets, without topoh modifications. But in dump from network interface by wireshark i see that packets was modified... -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com SATB-RIPE SATB-UANIC tel: (+38067) 403-62-54

2 17

rtpengine graphite statistics server
by Juha Heinanen 15 Jan '16

15 Jan '16

rtpengine README.md mentions "graphite statistics server". is that server openly available somewhere? -- juha

3 3

Kamailio and NAT
by Nelson Migliaro 15 Jan '16

15 Jan '16

Hello, I am running Kamailio behind NAT. Kanailio has a private IP and I am relaying NAT to internet router. I am using: - #!define WITH_NAT - listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060 - Patched RTP proxy including the advertise option And everything goes fine. I can make calls and have two way audio. The problem begins when the callee ends the call. BYE is not received in Kamailio (caller) I included the public IP using "add_contact_alias" because "set_contact_alias" was not adding the public IP. I included this in in NATDETECT (pre loaded router) if(is_first_hop()) { xlog("L_NOTICE","Metodo: $rm \n"); xlog("L_NOTICE","is first hop\n"); #set_contact_alias(); if (!add_contact_alias("PUBLIC-IP", "$Rp", "udp")) { xlog("L_ERR", "Error in aliasing contact $ct\n"); send_reply("400", "Bad request"); exit; } } I think the problem is related to destination that BYE is sent by the vendor. From what I see IP and port is taken from advertised in contact (PUBLIC-IP and 5060). The problem is that internet router changes the source port. Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1> --- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp", "udp") in order to received new transactions or should I follow a different procedure??? Thank you

4 14

dispatcher state x and the return of the probing mode
by Sven Neuhaus 14 Jan '16

14 Jan '16

Hello, when trying to set a destination to inactive using dispatcher.set_state after setting the state to "i", when I do a "dispatcher.list" I see the state as "ix". The "x" state appears not to be documented. After a short time (seconds) the flags return to state "ip" (probing). According to the module documentation I should be able to set the "probing" mode. 1. Why doesn't probing mode remain disabled? 2. What is state "x"? Cheers, -Sven

3 4

using re.subst with an $avp or $var
by Vik Killa 13 Jan '16

13 Jan '16

Hello, Is it possible to use $avp() or $var() inside re.subst? Example: $var(PrefixMatch) = "00"; $var(destnumber1) = $(ru{re.subst,/^sip:$var(PrefixMatch)(.*)@(.*)/\1/}); $var(destnumber2) = $(ru{re.subst,/^sip:00(.*)@(.*)/\1/}); xlog("L_INFO", "destnumber1 $var(destnumber1)\n"); xlog("L_INFO", "destnumber2 $var(destnumber2)\n"); In the above, destnumber2 works, yet destnumber1 does not. Thanks! /V

3 13

RTP Engine multiplexing and De-multiplexing
by riko nir 13 Jan '16

13 Jan '16

Hello, A call from a remote webrtc client is coming to (opensips+rtpengine). The media streams from the webrtc client is multiplexed. Can I use rtpengine to demultiplex the multiplexed streams and send it to other end as de-multiplexed SRTP traffic . This is because, the other end handle the srtp traffic only in demultiplxed manner. Also, the streams from the remote end is a separate SRTP and SRTCP traffic and the rtpengine has to add required ICE information to make the streams to go in a single port and it has to multiplex the streams and send it to webrtc client. It is mentioned that rtpengine can support multiplex/demultiplex. So, how can I configure rtpengine for this purpose from opensips? Your suggestions would be very helpful for us. Thanks. Riko

2 1

Regarding rtpengine dtls handling and fetching crypto keys
by riko nir 13 Jan '16

13 Jan '16

Hi all, I have a media server and it is able to handle SRTP, provided the crypto key. We are planning to give webrtc support to the media server. We are using opensips+rtpengine for that. For dtls, we are using rtpengine. The rtpengine just needs to do the dtls handshake and it needs to fetch the crypto key and it should provide the key to media server, so that the media server will be able to handle the SRTP traffic. We are bit struggling, how to get the crypto keys from rtpengine and send it to opensips and opensips will send it to media server. how this can be done? your suggestions will help us a lot. Thanks, Riko

2 4

siptrace retransmission's
by Grant Bagdasarian 12 Jan '16

12 Jan '16

Hello All, I'm load testing Kamailio with siptrace (duplicate) enabled, but for some reason not all SIP messages are arriving at the sip capture server (kamailio instance with sipcapture). The load is an average of 50 CPS. I enabled debug 9 and I get the following messages. <core> [mem/q_malloc.c:369]: qm_malloc(): qm_malloc(0x7fefde4da010, 1274) called from siptrace: siptrace.c: trace_send_hep_duplicate(1615) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde618a88), called from siptrace: siptrace.c: trace_send_hep_duplicate(1659) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde61ee18), called from siptrace: siptrace.c: trace_send_hep_duplicate(1660) <core> [mem/q_malloc.c:474]: qm_free(): qm_free: freeing frag. 0x7fefde61ede8 alloc'ed from siptrace: siptrace.c: trace_send_hep_duplicate(1615) <core> [mem/q_malloc.c:369]: qm_malloc(): qm_malloc(0x7fefde4da010, 1271) called from siptrace: siptrace.c: trace_send_hep_duplicate(1615) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde61b488), called from siptrace: siptrace.c: trace_send_hep_duplicate(1659) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde61e880), called from siptrace: siptrace.c: trace_send_hep_duplicate(1660) <core> [mem/q_malloc.c:474]: qm_free(): qm_free: freeing frag. 0x7fefde61e850 alloc'ed from siptrace: siptrace.c: trace_send_hep_duplicate(1615) DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission In siptrace.c I see the following line of code: if(ps->flags&TMCB_RETR_F) { LM_DBG("retransmission\n"); return; } Does this in fact retransmit the message to the capture server? Has anyone else load tested kamailio with siptrace on? I'm using Kamailio 4.1. I'm also using the setflag(..) function in siptrace module to duplicate all forwarded messages to the capture server. Regards, Grant

2 3

Panning next major release - v4.4
by Daniel-Constantin Mierla 11 Jan '16

11 Jan '16

Hello, sketching the road to the next major release, so people can plan their goals for it and discuss, if needed, before the start of winter holidays: - development freezing by end of January 2016 - test for one to one and a half month - release in the first part of March 2016 If there are important topics to decide on, we can organize an IRC meeting sometime in January 2016. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

4 10

behaviour of exit();
by Grant Bagdasarian 11 Jan '16

11 Jan '16

Hello, I have the following statements in my config script. When either of the IF statements is true, what does the exit function do? Will it still pass the message to the next hop or will it just abort and drop the sip message? failure_route[DISPATCH_FAILURE] { xlog("L_INFO", "[R-FAILURE-ROUTE-DISPATCH-FAILURE:$ci] !>" "Route:DISPATCH_FAILURE;UnixTime:$TV(un);RequestURI:$rU;RequestMethod:$rm;Status:$rs;FromTag:$ft;ToTag:$tt;CallID:$ci;From:$fU;To:$rU\r\n"); t_on_branch("MSG_MOD"); if(t_is_expired()) { xlog("L_INFO", "[R-FAILURE-ROUTE-DISPATCH-FAILURE:$ci] !>" "Transaction expired - aborting\r\n"); exit; } if (t_is_canceled()) { xlog("L_INFO", "[R-FAILURE-ROUTE-DISPATCH-FAILURE:$ci] !>" "Transaction cancelled - aborting\r\n"); exit; } Thanks. Regards, Grant

2 2

Ways to reload Kamailio configuration file without restart
by Amit Patkar 11 Jan '16

11 Jan '16

Hi What are different ways to reload Kamailio configuration file without restart? I need to make configuration file changes on production server without bringing down kamailio service. *Thanks & Regards,* Amit Patkar

2 3

Re: [SR-Users] RTPPorxy -> RTPEngine migration issue
by Arsen Hovhanissian 10 Jan '16

10 Jan '16

Richard that was a champion’s answer thank you so much! Maybe it should be noted kamailio's rtpengine module documentation.

1 0

Re: [SR-Users] RTPPorxy -> RTPEngine migration issue
by Arsen Hovhanissian 10 Jan '16

10 Jan '16

Hi Richard! Here’s the output at log level 7 [1452404737.610730] WARNING: Failed to properly parse UDP command line '11683_0 d7:command4:pinge' from 10.0.0.10:54602, using fallback RE [1452404737.619006] WARNING: Failed to properly parse UDP command line '11683_1 d7:command4:pinge' from 10.0.0.10:59567, using fallback RE [1452404737.621995] WARNING: Failed to properly parse UDP command line '11694_2 d7:command4:pinge' from 10.0.0.10:57256, using fallback RE [1452404737.623204] WARNING: Failed to properly parse UDP command line '11689_1 d7:command4:pinge' from 10.0.0.10:52176, using fallback RE [1452404737.629991] WARNING: Failed to properly parse UDP command line '11687_1 d7:command4:pinge' from 10.0.0.10:55951, using fallback RE [1452404737.630268] WARNING: Failed to properly parse UDP command line '11686_1 d7:command4:pinge' from 10.0.0.10:51267, using fallback RE [1452404737.630392] WARNING: Failed to properly parse UDP command line '11690_1 d7:command4:pinge' from 10.0.0.10:55452, using fallback RE [1452404737.634052] WARNING: Failed to properly parse UDP command line '11688_1 d7:command4:pinge' from 10.0.0.10:49329, using fallback RE [1452404737.635731] WARNING: Failed to properly parse UDP command line '11693_1 d7:command4:pinge' from 10.0.0.10:43723, using fallback RE [1452404737.636118] WARNING: Failed to properly parse UDP command line '11685_1 d7:command4:pinge' from 10.0.0.10:55434, using fallback RE [1452404737.636293] WARNING: Failed to properly parse UDP command line '11692_1 d7:command4:pinge' from 10.0.0.10:34893, using fallback RE Could it be that I am starting it up wrong? Thanks a million!

2 1

RTPPorxy -> RTPEngine migration issue
by Arsen Hovhanissian 10 Jan '16

10 Jan '16

Hi everyone, i’ve been having this “not so problem” going on. So I have rtpengine installed on a server and use the default rtpproxy module on kamailio and it works beautifully. Having read the rtpengine modules description, I see that it is a drop in replacement of rtpproxy. So I keep trying to update the module names and function names but it keeps giving me this message when I start up kamailio. ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid response Software versions: Kamailio: 4.3.4 RTPEngine: 4.1.1 Any ideas on what it could be?

2 1

asterisk server as SBC
by Muteesa Fred 09 Jan '16

09 Jan '16

Hello Daniel,Just got this email from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb… requirement is that I am looking at creating an asterisk server to work as SBC.wondering if you can help me on this one. thanks and regards,Fred

1 0

Kamailio - Asterisk: Handling loss of SIP BYE and dangling channels
by Benjamin Fitzgerald 08 Jan '16

08 Jan '16

Hi, I'm wondering what the best approach to handling a SIP dialog when one endpoint disappears/fails to send the BYE message. I have Kamailio as a proxy for all mobile (iPhone/Android) SIP clients. Occasionally, the user hangs up the call but no BYE message is received. This means that Asterisk has an open channel even though there is no client. Kamailio also continues to receive successful registrations from the SIP client so the endpoint is not down completely. Is Kamailio the appropriate place to handle this situation? What do you recommend? If not could you point me in the right direction? RTP timeout? Asterisk? The SIP client itself? Thanks for your help. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us

3 6

Audio issue when using 2 port ATA
by Daniel W. Graham 08 Jan '16

08 Jan '16

Setup is - 2 port ATA <> FIREWALL <> KAMAILIO / RTPPROXY <> ASTERISK If I have a single port in use behind the firewall, all NAT functions work properly and media is relayed through rtpproxy. If I have both ports in use behind the firewall, when outbound calls from UA are placed there is two way audio on both calls. However if inbound calls are placed to UA, the first call works, second call only has outbound audio. Different SIP URI is used for each port. If the firewall is eliminated everything works fine. Anyone have an idea how to troubleshoot or what could be missing? I have done packet captures on both the UA side and Kamailio side, and I see two RTP flows (rtp ports match on both sides as well) despite lack of inbound audio on the second call. If I can post anything config wise that would help let me know. Thanks! -Dan

2 11

Example kamailio.cfg t_check_trans()
by Phil Lavin 08 Jan '16

08 Jan '16

Hi there, The example config that Kamailio ships with contains this block of code in request_route: # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); I am not fully understanding the requirement for either of the t_check_trans() here. As far as I can see from the documentation, t_check_trans() returns a boolean value and does not modify any data internally. Can anyone help me find out what the relevance of the t_check_trans() calls is? Thanks Phil Lavin Systems Engineer CloudCall by SYNETY www.cloudcall.com<http://www.cloudcall.com> T: +44 (0) 330 335 0000 / +1 617 982 1600 D: +44 (0) 116 424 4790 / +1 617 982 4790 SM: LinkedIn<https://uk.linkedin.com/pub/phil-lavin/25/422/750> READ OUR BLOG FOR SMARTER COMMUNICATIONS<http://t.sidekickopen03.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XX43…> Confidentiality: This e-mail transmission, including any attachments, is intended only for the named recipient(s) and may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. If you have received this transmission in error, or are not the named recipient(s), please notify the sender immediately by return e-mail and permanently delete this transmission, including any attachments. Security: This e-mail and any attachments are believed to be free from any virus but it is the responsibility of the recipient to ensure this is so. E-mail is not a 100% secure communication.

2 3

using pipelimit at CPS (1 second)
by Vik Killa 08 Jan '16

08 Jan '16

Hello, has anyone used pipelimit at a timer_interval of 1 second? We would like to do rate-limiting based on CPS? The doc says it can be CPU intensive. Testing with SIPP, we see the CPU go up to around 30% at 500 CPS (with pipelimit blocking at 300 CPS) Thanks /V

3 4

Add Record-Route on 200 OK
by Igor Potjevlesch 08 Jan '16

08 Jan '16

Hello, I'm looking for a way to add a Record-Route into a 200 OK before relaying the message to the UA. Indeed, I'm facing an issue where the 200 OK is relayed to the UA without Record-Route and with an IP address into the Contact header. So, the UA sends back his ACK to the Contact instead of the proxy itself. Is it possible to achieve this? Regards, Igor.

3 10

config file error for kamailio wheezy
by Ramya Y 07 Jan '16

07 Jan '16

Hi, *I recently installed kamailio 4.1 on debian wheezy (i386)* Below are the list of modules in debian wheezy. root@debian:/etc/kamailio# find / -name modules /etc/modules /etc/initramfs-tools/modules /etc/gnome-vfs-2.0/modules /lib/modules /usr/share/initramfs-tools/modules /usr/share/doc/initramfs-tools/examples/modules /usr/share/doc/kamailio/modules /usr/lib/vmware-tools/lib64/libconf/pango/1.6.0/modules /usr/lib/vmware-tools/lib64/libconf/gtk-2.0/modules /usr/lib/vmware-tools/lib32/libconf/pango/1.6.0/modules /usr/lib/vmware-tools/lib32/libconf/gtk-2.0/modules /usr/lib/i386-linux-gnu/libproxy/0.3.1/modules /usr/lib/i386-linux-gnu/pango/1.6.0/modules /usr/lib/i386-linux-gnu/kamailio/modules /usr/lib/i386-linux-gnu/gio/modules /usr/lib/xorg/modules /usr/lib/gnome-vfs-2.0/modules /proc/asound/modules /proc/modules so i have added below mpath in kamaiio.cfg ,still i am facing errors .could you please help on it . *Added mpath in kamialio.cfg :* mpath="/.... .../:/usr/lib/i386-linux-gnu/kamailio/modules/" mpath="/usr/lib/i386-linux-gnu/kamailio/modules/" *facing error:* root@debian:/etc/kamailio# kamailio start loading modules under config path: /.... .../:/usr/lib/i386-linux-gnu/kamailio/modules/ loading modules under config path: /usr/lib/i386-linux-gnu/kamailio/modules/ ERROR: bad config file (1 errors) Segmentation fault note: For same debain wheezy used for PCSCF ,it is working fine . For ICSCF and SCSCF iam facing errors. only observed one is, for these i have installed mysql where for pcscf mysql is not installed. Is this any problem for modules. could you please assist me further. Thanks in Advance! Regards, Ramyasri.

3 5

Kamailio crashing after comparing AVP with string
by Jurijs Ivolga 07 Jan '16

07 Jan '16

Hi, I have weird behavior, when I'm trying to compare AVP with string: if ($avp(AVP_ATTRS) != "bla") { ds_next_dst(); } $avp(AVP_ATTRS) contains Dispatcher attribute. Kamailio crashes: Dec 17 03:39:30 /usr/local/sbin/kamailio[4844]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 15 Dec 17 03:39:30 /usr/local/sbin/kamailio[4810]: ALERT: <core> [main.c:728]: handle_sigs(): child process 4816 exited by a signal 11 Dec 17 03:39:30 /usr/local/sbin/kamailio[4810]: ALERT: <core> [main.c:731]: handle_sigs(): core was not generated Dec 17 03:39:30 /usr/local/sbin/kamailio[4810]: INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD Dec 17 03:39:30 /usr/local/sbin/kamailio[4843]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4842]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4841]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4840]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4839]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4838]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4837]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4836]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4835]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4834]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4829]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4814]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4815]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4823]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4813]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4821]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4812]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4818]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4825]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4819]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4822]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4844]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Dec 17 03:39:30 /usr/local/sbin/kamailio[4830]: INFO: <core> [main.c:798]: sig_usr(): signal 15 received Following code works fine: if ($avp(AVP_ATTRS) == "bla") { ds_next_dst(); } Kamailio version: version: kamailio 4.3.4 (x86_64/linux) a56227 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: a56227 compiled on 03:29:23 Dec 17 2015 with gcc 4.4.7 Thak you! With kind regards, Jurijs

2 5

Negative value returned when using sql_pvquery
by Jonathan Hunter 06 Jan '16

06 Jan '16

Hi Guys, Wonder if anyone has experienced this before, on kamailio 4.1. I am using an sql_pvquery and it works fine, unless I pull a value from the database that is defined as 10 digits and starts with a 3, actually its populated as 03785702370 but due to the table it removes the leading 0 (bigint(20) unsigned). Running a manual query in my database for a DestinationMsisdn I get; select DestinationMsisdn,SourceMsisdn from MsisdnPoolAllocations where PoolMsisdn='+44162489xxxx';+-------------------+--------------+| DestinationMsisdn | SourceMsisdn |+-------------------+--------------+| 3785702370 | 4474526xxxxx |+-------------------+--------------+ However running in the kamailio.cfg an sql_pvquery to do the same query as below; sql_pvquery("cd","select DestinationMsisdn,SourceMsisdn from MsisdnPoolAllocations where PoolMsisdn='$rU'","$var(MOdest),$var(NewSourceMSISDN)"); However this returns a value of -509264926 for $var(MOdest) which should just be the 3785702370 number. What can cause kamailio to interpret this as a negative value? Has anyone seen this before? Many thanks Jon

3 6

Re: [SR-Users] Fosdem 2016
by DanB 06 Jan '16

06 Jan '16

Hey Daniel, Please also count me in (and possible 2 other friends) for dinner/event out. Thanks, DanB

1 0

Kamailio World 2016 – Call For Speakers
by Daniel-Constantin Mierla 06 Jan '16

06 Jan '16

Hello, should anyone here be interested in submitting a proposal for a presentation at Kamailio World 2016, note that the process is open now. More details were posted on a news article at: - http://www.kamailio.org/w/2016/01/kamailio-world-2016-call-for-speakers/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Kamailio Password Encryption
by micho fr 05 Jan '16

05 Jan '16

Hi Guys, I have a kamailio server 1.3 with clear text Password in Subscriber table.... I wish I can encrypt that password by doing some kind of combinations between username, domain and let kamailio server be reading the encrypted password Is there a way to do it? modparam("auth_db", "db_url", DBURL)modparam("auth_db", "calculate_ha1", 0)modparam("auth_db", "password_column", "password")#modparam("auth_db", "password_column", "ha1")modparam("auth_db", "password_column_2", "ha1b")modparam("auth_db", "load_credentials", "")modparam("auth_db", "use_domain", MULTIDOMAIN)

3 2

(no subject)
by Safdar Khan 05 Jan '16

05 Jan '16

I have made a calling service through kamailio and rtpengine.In this setup of calling service i have also included webRTC for cross platform(like android phone to web browser).Same calling service is not working after addition of freeswitch.Additional integration of Freeswitch in this current setup is making trouble. I have followed below link to setup my rtpengine module. https://github.com/caruizdiaz/kamailio-ws <https://github.com/caruizdiaz/kamailio-ws> My requirement is to integrate all these three to make cross platform calling system Kamailio<-->Rtpengine<-->Freeswitch But the calls are not relaying to freeswitch via rtpengine. Help me out to relay calls from rtpengine to freeswitch in this setup. Thanks and Regard Md Safdar Khan

2 1

Re: [SR-Users] Parser error when enabling siptrace with HEP functionality
by Sergey Basov 05 Jan '16

05 Jan '16

Thanks for reply. I check one more time, and does not see any non SIP (HEP) packets received on port 5060 of kamailio. May bee parser trying to parse (HEP) packet when it going out via socket to capture server? When I turn of topoh module I does not see any errors in log. And in both cases I see HEP packets on kamailio capture server. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com SATB-RIPE SATB-UANIC tel: (+38067) 403-62-54

2 1

Configure Kamailio to generate OPTIONS
by Mititelu Stefan 05 Jan '16

05 Jan '16

Hello, What is the simplest way to configure Kamailio to generate periodical OPTIONS to the caller, for an established dialog? Can you give me some starting references? Thanks, Stefan

4 4

Is it possible to use kamailio as a Webrtc gateway and integrate it into to a conference bridge for handling webrtc clients in conference bridge?
by suganthi karthick 05 Jan '16

05 Jan '16

Hi all, I need to implement a WebRTC gateway for an existing conference bridge. The WebRTC gateway has to support Signaling, ICE and DTLS. The webrtc clients can be JsSIP or any webrtc client. The conference bridge is an existing working one for SIP clients, and I am trying to add webrtc support for that. The webrtc gateway needs to be implemented in a way like a library because it needs to be integrated into the existing platform. There are some init functions and config function from the existing conference platform, based on which the webrtc gateway has to be configured. Also, when a webrtc call come from a webrtc client, it needs to handle the signaling and the media(RTP) has to go to the conference bridge platform. It would be really helpful if you suggest whether I can use openSIPS for this purpose and use it as a library and integrate into the exiting platform? Your suggestions will be more helpful. Thanks.

2 1

Parser error when enabling siptrace with HEP functionality
by Sergey Basov 05 Jan '16

05 Jan '16

Hi, All. I have a strange issue occuring, when I enable the siptrace functionality within the routing logic, I begin to get a lot of parser error as show below. Is this normal behavior? This errors appears only when topoh module is enabled... As I see with wireshark, there is no duplicates or returned HEP packets... I enable siptrace as follows: request_route { sip_trace(); setflag(22); ... } Configuration of the topoh and siptrace modules: # ----- topoh params ----- modparam("topoh", "mask_key", "123456789") modparam("topoh", "mask_ip", "127.0.0.8") #Siptrace modparam("siptrace", "duplicate_uri", "sip:10.1.23.20:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) modparam("siptrace", "force_send_sock", "sip:10.1.23.23:5060") modparam("siptrace", "traced_user_avp", "$avp(s:user)") output from kamailio.log with debug enabled: Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [th_msg.c:180]: th_mask_via(): body: 120: [SIP/2.0/UDP 10.10.206.39:5060 ;received=10.10.206.39;TH=dcv;branch=z9hG4bK-d8754z-2e9df22b7c5cb6c9-1---d8754z-;rport=5060] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [th_msg.c:191]: th_mask_via(): +body: 199: [SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-s7wTDLa0zUfYZXl5zpl0zpl0z.lRD.zok.sEG.lvBJY.euwReuTfzpl0zpl0z.lRD.zokqZaxuZ.P.c3BJX01R5fO.wCZLZ3WSqMkhBqGgCczJsoeV1SzJa21LY.1.e.kW7jDW7ceh52GpZKDpcSBVfSPh7qzh1E] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [th_msg.c:322]: th_mask_record_route(): no record route header Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: tm [t_hooks.c:266]: run_trans_callbacks_internal(): DBG: trans=0x7fcfacccb7a8, callback type 4194304, id 0 entered Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:766]: sip_trace_store_db(): database connection not initialized Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1875]: pipport2su(): the port string is 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1875]: pipport2su(): the port string is 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [proxy.c:265]: mk_proxy(): doing DNS lookup... Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [resolve.c:1208]: srv_sip_resolvehost(): 10.1.23.20:9060 proto=1 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [resolve.c:1329]: srv_sip_resolvehost(): returning 0x9da020 (10.1.23.20:9060 proto=1) Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1687]: trace_send_hep_duplicate(): setting up the socket_info Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1694]: trace_send_hep_duplicate(): force_send_sock activated, grep for the sock_info Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==11 && [10.1.23.23] == [10.56.41.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==11 && [10.1.23.23] == [10.56.42.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.1.23.23] == [10.1.23.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: siptrace [siptrace.c:1701]: trace_send_hep_duplicate(): found socket while grep: [10.1.23.23] [10.1.23.23] Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: <core> [parser/msg_parser.c:602]: parse_msg(): parse_msg: invalid message Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023�#023�#0128*#027#0128*#024zx�V�#016#003> Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: topoh [topoh_mod.c:214]: th_prepare_msg(): outbuf buffer parsing failed! Jan 5 10:02:02 sip1 /usr/sbin/kamailio[28348]: DEBUG: tm [t_funcs.c:362]: t_relay_to(): SER: new transaction fwd'ed Is there any way I can disable seeing these messages? What are these messages? Any thoughts are appreciated. Thanks in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 1

Kamailio-IMS icscf erro
by Kranti Kumar 05 Jan '16

05 Jan '16

Hi, We are using attached two configuration file to start icscf node. But while we run "kamailio start" command getting below errors. loading modules under config path: /usr/lib/i386-linux-gnu/kamailio/modules/ ERROR: bad config file (1 errors) Segmentation fault Can anyone help me here to resolve this issue. Regards, -kranti

3 3

Too many packets in rtpengine UDP receive queue
by Zodiac 04 Jan '16

04 Jan '16

Hi friends: I am running rtpengine daemon on a CentOS machine functionally. SIP server is Kamailio. Vedio and audio calls are both OK with rtpengine. But there is always prompts like following in rtpengine’s log: Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30136] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30156] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible Dec 22 19:57:52 localhost rtpengine[12679]: [oaNEqGokmqgaXzj0xWyPeJDGPX7ln0gG port 30156] Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible So I am wondering whether this prompt is normal. Otherwise, what should I do to prevent this to happen? Thx a lot! ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

3 4

Re: [SR-Users] Presence - Subscription based NOTIFY to enable MWI
by Arsen Hovhanissian 03 Jan '16

03 Jan '16

Alright I guess this is one of those duhh… moments, all this time I was forgetting to add the content length to the NOTIFY event. Remark: DO NOT FORGET the content length! Thanks for the help, very much appreciated!

1 0

Re: [SR-Users] Presence - Subscription based NOTIFY to enable MWI
by Arsen Hovhanissian 03 Jan '16

03 Jan '16

I see, what i’m trying to do is actually to limit querying the DB by updating the MWI only when there is activity. In other words to send a NOTIFY when the user receives a new message in his voicemailbox. I am not sure if it’s possible to send the NOTIFY for example 5-30 mins after the initial subscribe.

2 1

Presence - Subscription based NOTIFY to enable MWI
by Arsen Hovhanissian 03 Jan '16

03 Jan '16

Hi everyone, I’m trying to send a NOTIFY event using “sipsak” to enable the MWI I read up a lot of documentation and didn’t really find the information needed to accomplish this. So I have: Phone 1 -> P1 Server 1 -> S1 (Kamailio 4.3.4) Server 2 -> S2 (sipsak) P1 registers to S1 and creates a new entry in the active watchers table with a "message-summary” event. At this point I am assuming that any request I would send will be out of dialog. Now the question is, How could I reply to that SUBSCRIPTION with a NOTIFY? Thanks in advance!

2 1

Happy New Year!
by Daniel-Constantin Mierla 31 Dec '15

31 Dec '15

The amazing 2015 for Kamailio is done! Thank you everyone for contributing to it! Now it is the time to get prepared to fill the 2016, with interesting discussions on our forums, solving new RTC challenges, adding features in Kamailio and welcoming new contributors! I wish a healthy and prosperous year to all Kamailio friends! I am looking forward to meeting many of you at Kamailio World Conference, where we are going to celebrate 15 years of development for Kamailio project, as well as at other events around the globe! Happy New Year! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda http://www.kamailioworld.com

2 1

tsilo and linphone android connected via TCP
by Vasiliy Ganchev 29 Dec '15

29 Dec '15

Hi folk! sorry for first sending to the wrong list ! Currently playing around with push notification in kamailio, have a bad case. Would appreciate any idea. Scenario (all devices connected via TCP): UA-A - sipphone, UA-B[1] - sipphone UA-B[2] - Linphone android Server - kamailio+asterisk UA-A call UA-B (UA-B[1] - online, UA-B[2] - offline) Kamailio make ts_store, and t_relay to UA-B[1], for UA-B[2] kamailio send push notification UA-B[2] appear registered, I make ts_append, and see kamailio sending INVITE to UA-B[2], but tansport used is UDP (expected TCP) If I use UDP on UA-B[2] - have no issue. I am using kamailio-4.2.5, looked in sources of current master, did not found changes that could help in my scenario. General question - does anyone has same problem using ts_append for TCP connected devices? If TCP in ts_append should work OK, any other idea? cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/tsilo-and-linphone-android-connecte… Sent from the Users mailing list archive at Nabble.com.

2 2

kamailio wheezy config files
by Ramya Y 29 Dec '15

29 Dec '15

Hi , Greetings! i have configured mpath in kamailio.cfg ,when i run kamailio component of icscf (wheezy). i am facing below Errors. *error:* root@debian:/etc/kamailio# kamailio start root@debian:/etc/init.d# kamailio start loading modules under config path: /usr/lib/kamailio/modules:/usr/lib/kamailio/modules_s:/usr/lib/kamailio/modules_K loading modules under config path: usr/local/ims-kamailio/kamailio-4.2.0/output/lib64/kamailio/modules/ loading modules under config path: /.... .../:/usr/lib/i386-linux-gnu/kamailio/modules/ ERROR: bad config file (1 errors) Segmentation fault could you please suggest me which mpath i have to configured in kamailio.cfg for wheezy. Thanks in Advance! Regards, Ramyasri

2 1

Append to reply in default action of failure_route
by Efelin Novak 28 Dec '15

28 Dec '15

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

3 3

notify takes several seconds to generate
by Juha Heinanen 28 Dec '15

28 Dec '15

i noticed that kamailio 4.3 presence server takes several seconds to generate notify when it receives subscribe request. for example: Dec 27 15:42:01 lohi /usr/bin/sip-proxy[7363]: INFO: Routing SUBSCRIBE <sip:test@test.tutpro.com> to presence server at <sip:127.0.0.1:5080;transport=tcp> Dec 27 15:42:01 lohi /usr/bin/pres-serv[7113]: INFO: Handled SUBSCRIBE <sip:test@test.tutpro.com> Dec 27 15:42:05 lohi /usr/bin/pres-serv[7108]: INFO: Routing locally generated NOTIFY <sip:test@test.tutpro.com;gr=urn:uuid:47759291-c9db-4eb3-a540-5e68353414ee> to <sip:test@test.tutpro.com> from <sip:test@test.tutpro.com> the host is not loaded at all. these relevant presence server params are set: modparam("presence", "subs_db_mode", 3) modparam("presence", "db_table_lock_type", 0) and db is on localhost. any explanation why notify takes so long to generate? -- juha

2 3

how to make secure websocket run on kamalio
by tintuc tv 28 Dec '15

28 Dec '15

Hi list is there any tutorial for making secure websocket work with kamailio. the SSL Certificates is bought from SSL Certificates thank you

2 1

kamailio 4.0.1 websocket configuration
by Rupayan Dutta 28 Dec '15

28 Dec '15

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

4 3

Best way to store RTPProxy statistics in database
by Gholamreza Sabery 28 Dec '15

28 Dec '15

I want to store RTPProxy statistics in $rtpstat variable (exported by RTPProxy module) into database. Currently I use app_python module to run a specific Python script with from_tag, to_tag, callid and $rtpstat as parameters. This module puts this information in a message queue which is red by another process. That process writes the data in the database. Is there a better, more efficient way to store arbitrary data like this into database in Kamailio? Regards

2 2

Asterisk failover
by Phil Daws 28 Dec '15

28 Dec '15

Seasons Greetings! Am starting to learn about Kamailio fronting Asterisk and wish to ask how one performs multi-site failover. What I would like to be able to do is when the upstream trunk provider routes a SIP call to me if Kamailio detects that the backend Asterisk is down it will send to a secondary system ? Thanks, Phil

2 1

Too many packets in rtpengine UDP receive queue
by Zodiac 27 Dec '15

27 Dec '15

Hi friends: I am running rtpengine daemon on a CentOS machine functionally. SIP server is Kamailio. Vedio and audio calls are both OK with rtpengine. But there is always prompts like following in rtpengine’s log: So I am wondering whether this prompt is normal. Otherwise, what should I do to prevent this to happen? Thx a lot! ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

2 6

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 25 Dec '15

25 Dec '15

With an amazing 2015 almost gone, I am using this moment to give my thanks and greetings to the people involved in Kamailio project, old and new friends, developers, contributors, the engaged and warm community members. Very soon 2016 will arrive, Kamailio will celebrate 15 years of development, therefore we are looking to a special year ahead! Merry Christmas and Happy Winter Holidays! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

AUTH doesn't work
by Evgeny Zhavoronkov 23 Dec '15

23 Dec '15

Hello! Here is my config: http://pastebin.com/X20JumYB Why can't I authorize? I get: 403 Not relaying (kamailio (4.3.4 (x86_64/linux))) Interesting that with previous version everything was ok. I just use my previous config. -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

2 2

Issue with end to end MESSAGE over TCP
by Thierry Luo 23 Dec '15

23 Dec '15

Hi All, I have deployed a Kamailio proxy server (as IMS pCSCF) in Amazon EC2. UA registration over TCP and presence flow (PUBLISH/NOTIFY) are all fine. But MESSAGE from one UA to another is UA over TCP is always not successful. I captured the sip flow from the server, and found that at the final hop from Proxy to terminating UA, the Kamailio server send the MESSAGE over udp, regardless of the fact that UAs were registered over TCP. What is the problem maybe? Attached is the cfg file. Thanks. Thierry Luo

2 4

rtpproxy does not response, disable it
by Zodiac 21 Dec '15

21 Dec '15

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

1 0

lost mysql connections
by Juha Heinanen 21 Dec '15

21 Dec '15

for some reason once in a while kamailio processes fail to execute mysql queries while at the same time an mysql test script that is run once per second from cron works fine. kamailio executes mysql queries during normal sip message processing and also when it processes an OPTIONS ping test request: sql_query("sip_proxy", "SELECT table_version FROM version WHERE table_name='domain'", "result"); if ($rc == 1) { sql_result_free("result"); options_reply(); } else { xlog("L_ERR", "$rm <$ru> failed sql test\n"); sql_result_free("result"); send_reply("500", "Server Internal Error"); }; here is an example: ... Dec 15 00:51:08 proxy1 php: MySQL server test succeeded Dec 15 00:51:09 proxy1 php: MySQL server test succeeded Dec 15 00:51:10 proxy1 /usr/bin/sip-proxy[29894]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query Dec 15 00:51:10 proxy1 /usr/bin/sip-proxy[29894]: ERROR: db_mysql [km_dbase.c:145]: db_mysql_async_exec_task(): failed to execute query on async worker Dec 15 00:51:10 proxy1 php: MySQL server test succeeded Dec 15 00:51:11 proxy1 php: MySQL server test succeeded Dec 15 00:51:13 proxy1 php: MySQL server test succeeded Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29891]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29891]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query Dec 15 00:51:13 proxy1 /usr/bin/sip-proxy[29883]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:14 proxy1 php: MySQL server test succeeded Dec 15 00:51:14 proxy1 /usr/bin/sip-proxy[29886]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:15 proxy1 php: MySQL server test succeeded Dec 15 00:51:15 proxy1 /usr/bin/sip-proxy[29885]: ERROR: OPTIONS <sip:xxx:5060> failed sql test Dec 15 00:51:16 proxy1 php: MySQL server test succeeded Dec 15 00:51:17 proxy1 php: MySQL server test succeeded Dec 15 00:51:18 proxy1 php: MySQL server test succeeded ... all db_mysql module params are at their default values. any idea why kamailio at connection loss does not automatically create new connections? is there a means to somehow debug this further? -- juha

2 3

Kamailio and Skype for Business (MS Lync)
by Grant Bagdasarian 21 Dec '15

21 Dec '15

Hello, Does anyone have any experience configuring Kamailio as a gateway for Skype For Business (Microsoft Lync)? Regards, Grant

3 2

ACC module from_tag, to_tag and callid
by Gholamreza Sabery 21 Dec '15

21 Dec '15

In the ACC modules' database table (acc) in Kamailio sometimes from_tag, to_tag and callid of an INVITE and corresponding BYE completely match. But sometimes from_tag in INVITE is to_tag of BYE and to_tag of BYE will become from_tag of INVITE. Why this happens? Is this normal? Regards

3 6

crash in version 4.0
by Juha Heinanen 21 Dec '15

21 Dec '15

core dump occurred in kamailio version 4.0. does anyone remember if a bug causing such a crash would possibly have been fixed in some later kamailio version? -- juha ------------------------------------------------------------------------------ Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21089]: WARNING: db_mysql [km_dbase.c:85]: driver error on ping: Lost connection to MySQL server during query Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21116]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 90 Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21016]: ALERT: <core> [main.c:788]: child process 21089 exited by a signal 11 Dec 17 17:16:59 node5 /usr/sbin/sip-proxy[21016]: ALERT: <core> [main.c:791]: core was generated ------------------------------------------------------------------------------ Program terminated with signal 11, Segmentation fault. #0 run_trans_callbacks_internal (cb_lst=<value optimized out>, type=512, trans=0x7f093593a9b0, params=0x7fff7a5289c0) at t_hooks.c:286 286 t_hooks.c: No such file or directory. in t_hooks.c (gdb) bt full #0 run_trans_callbacks_internal (cb_lst=<value optimized out>, type=512, trans=0x7f093593a9b0, params=0x7fff7a5289c0) at t_hooks.c:286 cbp = 0x3030 backup_from = 0x8a49d0 backup_to = 0x8a49d8 backup_dom_from = 0x8a49e0 backup_dom_to = 0x8a49e8 backup_uri_from = 0x8a49c0 backup_uri_to = 0x8a49c8 backup_xavps = 0x8a4ad0 #1 0x00007f0941ad3ddc in run_trans_callbacks_with_buf (type=1887007839, rbuf=<value optimized out>, req=<value optimized out>, repl=0x6565705f74736423, flags=0) at t_hooks.c:336 params = {req = 0x7f0935b32360, rpl = 0xffffffffffffffff, param = 0x7f0935940900, code = 408, flags = 408, branch = 0, t_rbuf = 0x7f093593aa70, dst = 0x7f093593aac0, send_buf = { s = 0x7f093594d180 "From: xxx <sip:xxx@xxx:5060;user=phone>;tag=262063958\r\nTo: xxx <sip:xxx@xxx:5060;user=phone>;tag=as2db5b848\r\nCall-ID: 2864df7e-1d31c032-6dd726"..., len = 462}} trans = 0x246 #2 0x00007f0941aef8cb in relay_reply (t=0x7f093593a9b0, p_msg=<value optimized out>, branch=<value optimized out>, msg_status=<value optimized out>, cancel_data=<value optimized out>, do_put_on_wait=<value optimized out>) at t_reply.c:1880 relay = 0 save_clone = 0 buf = 0x7f0943b3d140 "SIP/2.0 408 Request Timeout\r\nVia: SIP/2.0/UDP xxx:5060;branch=z9hG4bKterm-2b3acd-xxx-xxx-72778\r\nFrom: xxx <sip:xxx@xxx:5060;user=phone>;tag=17"... res_len = 462 relayed_code = 408 relayed_msg = 0xffffffffffffffff reply_bak = 0x13c400000050c7d3 bm = {to_tag_val = { s = 0x7f0943b3d259 "0b5775687262b458dc9c8d93773b9caa-d95c\r\nCall-ID: 1be7fd61-4cf116a3-62498d9c-8757@xxx\r\nCSeq: 1 INVITE\r\nServer: Kamailio (4.0.1 (x86_64/linux))\r\nContent-Length: 0\r\n\r\n", len = 37}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7f093593aa70 to_tag = <value optimized out> reason = {s = 0x56dbc5 "Request Timeout", len = 15} onsend_params = {req = 0x0, rpl = 0x1, param = 0x7f093593ab18, code = 898869680, flags = 32521, branch = 0, t_rbuf = 0x7d9c88, dst = 0xfffffe22, send_buf = { s = 0x7f093593ab20 "", len = 1101857122}} #3 0x00007f0941af5ac1 in fake_reply (t=0x7f093593a9b0, branch=0, code=<value optimized out>) at timer.c:354 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 898870080}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 898870080}}}} reply_status = <value optimized out> #4 0x00007f0941af5ec8 in final_response_handler (ticks=1904446327, tl=0x7f093593ab40, p=<value optimized out>) at timer.c:526 prev_branch = -478 now = 0 #5 retr_buf_handler (ticks=1904446327, tl=0x7f093593ab40, p=<value optimized out>) at timer.c:584 rbuf = 0xfffffe22 fr_remainder = <value optimized out> retr_remainder = <value optimized out> retr_interval = <value optimized out> new_retr_interval_ms = <value optimized out> #6 0x00000000004fa080 in slow_timer_main () at timer.c:1151 n = 12 ret = 12 tl = 0x7f093593ab40 #7 0x0000000000458039 in main_loop () at main.c:1693 i = 16 pid = <value optimized out> si = 0x0 si_desc = "sctp receiver child=15 sock=xxx:5060\000\000\000\200\215Rz\377\177\000\000\000\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000\001\000\000\000\004\000\000\000\005\000\000\000\000\000\000\000\020\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000 \251\212", '\000' <repeats 13 times>, "\020\000\000\000\000\000\000" nrprocs = 16 #8 0x00000000004599fb in main (argc=17, argv=<value optimized out>) at main.c:2566 cfg_stream = <value optimized out> c = <value optimized out> r = 0 tmp = 0x7fff7a529f11 "" tmp_len = 0 port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 2572917976 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7fff7a528d80 p = <value optimized out> (gdb)

2 1

rtpengine module rtpengine_disable_tout question
by Juha Heinanen 19 Dec '15

19 Dec '15

rtpengine readme tells: 4.2. rtpengine_disable_tout (integer) Once an RTP proxy was found unreachable and marked as disabled, the rtpengine module will not attempt to establish communication to that RTP proxy for rtpengine_disable_tout seconds. Default value is "60". does this mean that after 60 seconds is passed, rtpengine module will again try to use that RTP proxy, i.e., the RTP proxy does never become permanently disabled unless disabled by nh_enable_rtpp mi command? does the this param apply also if all RTP proxies of a set (perhaps consisting of only one proxy) are unreachable? -- juha

2 2

Kamailio v4.1.9 Released
by Daniel-Constantin Mierla 18 Dec '15

18 Dec '15

Hello, Kamailio SIP Server v4.1.9 stable release is out. This is marking the end of maintenance for 4.1.x series. Kamailio project maintains the last two stable branches, at this moment these are 4.2 and 4.3. Therefore, anyone running older releases is advised to upgrade in the near future to the latest version of one of those branches, preferably 4.3, in order to benefit of proper maintenance. For more details about version 4.1.9 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/12/kamailio-v4-1-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.4. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

websocket sometimes log errors after ws connect has closed
by Vasiliy Ganchev 18 Dec '15

18 Dec '15

Hi list! firstly, the described below issue is harmless to us, and is just little bit annoying, but analysing the nature of issue decided to share the results, and hear the opinion of community. Scenario: - WebRTC client connected to Kamailio's websocket - after some time I close tab with WebRTC, in browser. - using "kamctl fifo ws.dump" and "netstat -npa |grep kama |grep tcp" I can see that the connection is closed - but Kamailio still triing to send keepalive to that connection, and in syslog I got errors: Dec 18 12:28:18 /usr/sbin/kamailio[4471]: WARNING: websocket [ws_frame.c:245]: encode_and_send_ws_frame(): TCP/TLS connection get failed Dec 18 12:28:18 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_frame.c:797]: ping_pong(): sending keepalive Analysing sources, and using debug level 3 I see such logs (after the webrtc client is closed, and ) Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:468]: wsconn_get_list(): counter wsc [0xb1acb270] prev => [(nil)] next => [(nil)] Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:495]: wsconn_get_list(): wsc [0xb1acb270] id [2] ref++ Dec 18 12:33:33 /usr/sbin/kamailio[4471]: ERROR: websocket [ws_conn.c:504]: wsconn_get_list(): wsconn_get_list returns list [0xb666f410] with [1] members What is strange: in some tests, after unsuccessful sending of Ping, kamailio's "wsconn_get_list" clean up that connection, but in some tests, kamailio try sending Ping to old connection during 5-10 minutes (this time is not constant from test to test) Suddenly my current knowledge of C is not enough to find out were is the problem's root. General question is: is this correct list for such issuing? Or it should be in Dev list? once again - the problem is harmful (afetr some time, kamailio stop trying sending ping to inactive connection, but it is annoying to see errors in logs) Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/websocket-sometimes-log-errors-afte… Sent from the Users mailing list archive at Nabble.com.

1 1

issue with BYE
by gerry kernan 18 Dec '15

18 Dec '15

Hi I'm looking for any pointers to resolve and issue I have . When callee does a hang-up the call terminates on kamailio and asterisk but the caller's phone handset doesn't receive the BYE and doesn't hang-up. If the caller does the hang up then the call terminates and the callee's handset hands up . My setup is Kamailio WAN -> Kamailio Lan -> Asterisk LAN I'm using Path and dispatcher modules to route the SIP traffic to the backup end asterisk server my the domain in the SIP request. Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - www.infinityit.ie IP Telephony Asterisk Consulting - www.asteriskconsulting.com Contact Centre Total Interact - www.totalinteract.com

3 7

Releasing Kamailio v4.1.9
by Daniel-Constantin Mierla 18 Dec '15

18 Dec '15

Hello, I am going to release v4.1.9 later today, as last to be officially packaged version from branch 4.1. It is strongly recommended to migrate to a newer version, because the focus of developers is on the two most recent stable branches. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

kamailio as simple sip forwarder / gateway / proxy
by Peter Gervai 17 Dec '15

17 Dec '15

Hello, I've been browsing the net and the (not so friendly) documentation but found no readily available solution. I have tried to fabricate on myself but it seems it's not that easy without getting deeply familarised with SIP protocol internals which I don't quite want know. The problem seems simple enough, maybe you can easily tell the solution. There is kamailio at ip 256.10.10.1. It's task is to accept connections from SIP clients and forward them to PBX at 256.20.20.1, and accept PBX packets and forward them to the clients. PBX handles registration, options, invites, everything. Kamailio does not authenticate, authorise, just forwards everything back and forth, like a mindless proxy. No NAT involved. [The reason is that K. have to call external scripts whenever an outbound call (towards the clients) is initiated, which the PBX is unable to. Please don't advise other solutions (which isn't able to call executables in-session).] I have created a simplistic config which is not expected to work and it doesn't quite (plenty of security stuff left out now): route { if(src_ip==myself) { add_path(); $du="sip:256.20.20.1:95060"; } else { if(src_ip!="256.20.20.1") { $du="sip:256.20.20.1:95060"; } else { #from gateway pkts aren't changed } if(!t_relay()) { sl_reply_error(); break; } } There is several possible problems, but it seems Contact: is one main one since it contains the client instead of the proxy. Possibly this may even be the almost-good solution. I'm sure there's an easy and pretty solution so before I start to make ugly ones I ask the wise people. Thanks, Peter

2 1

Kamailio v4.2.7 Released
by Daniel-Constantin Mierla 17 Dec '15

17 Dec '15

Hello, Kamailio SIP Server v4.2.7 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.7 (or to 4.3.x series). For more details about version 4.2.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/12/kamailio-v4-2-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.4. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

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Kamailio as a SBC
by Guillaume 17 Dec '15

17 Dec '15

Hi everyone, I'm trying to make kamailio works as a SBC in front of Asterisk. I have read the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and it is close to what I'm looking for but still a little different. I want don't want kamailio to do auth and I want it to be a media gateway. The media and SIP path have to look like that: Phone 1 ----- kamailio -----Asterisk ---- Kamailio ---- Phone 2 First I have add an outboundproxy field in the Asterisk configuration to make all SIP messages from Asterisk passe through Kamailio. Then I started a minimal conf for Kamailio and it works at least for the SIP part. Call can be established perfectly. However the RTP proxy part is a mess. Kamailio changes the IP address in the sdp part only for INVITE messages and not for the 200 OK responses, so there is only one side of the media passing through Kamailio, the other side is sent directly to asterisk. But even for the half part rtpproxy craches with a segfault and Kamailio doesn't transmit this media. I have been searching some helps on google but nothing so far :/ I hope that someone here could help me. ----------------------------------------------- #!KAMAILIO ####### Defined Values ######### #!define ASTERISK 192.168.51.43 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=2 auto_aliases=no # to avoid simple attacks, use unconventional port port=5060 ####### Modules Section ######## mpath="/usr/local/lib/kamailio/modules" loadmodule "db_text.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "path.so" loadmodule "rtpproxy.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # 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) modparam("rr", "append_fromtag", 0) modparam("path", "use_received", 1) modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## route { xlog("L-ALERT", "new message !!!!!!!!!!!!!!!"); if (!sanity_check()) { exit; } if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); xlog("L_NOTICE", "$si $rm $ru -- too many hops\n"); exit; } if (af==INET) { force_rport(); if (src_ip != ASTERISK) { $du = "sip:192.168.51.43:5060"; add_path_received(); route(RTPPROXY); record_route(); } else { loose_route(); route(RTPPROXY); record_route(); } } forward(); } route[RTPPROXY] { rtpproxy_manage("cwei"); } onreply_route { if (af==INET) { rtpproxy_manage("cwei"); } return(1); } --------------------------------------------- Guillaume

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Planning to release v4.2.7
by Daniel-Constantin Mierla 17 Dec '15

17 Dec '15

Hello, later this week, on Thursday, Dec 17, I am planning to release a minor version based on branch 4.2, respectively 4.2.7 -- it's been a while since v4.2.6. As usual, if there are issues not yet reported or patches to be backported, write to sr-dev mailing list. If time allows, in the same day or next days, I plan to do the last release of branch 4.1 to mark the end of official packaging in 4.1 series. At this moment, the last two stable branches are 4.3 and 4.2. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 1

Presence Module
by Loïc Boissy 16 Dec '15

16 Dec '15

Hello, I use kamailio 4.3.2 with the presence module. When the contact is able to manage its presence, No problem for me. But when a contact doesn't manage it presence, i would like to know if it is possible to generate a sip message (like as a simple presence message) with kamailio? The goal of that is to prevent my watcher when a "simple" contact (without presence option) is registered and unregistered on my kamailio server. Thank you for your help. Best regards. Loïc BOISSY

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State-full failover Kamailio+RTPProxy
by Frank Costeira 16 Dec '15

16 Dec '15

Hi, Is there anyway to do state-full failover of Kamailio+RTPProxy from a primary server to a backup server? Regards, Frank

5 7

New logo for Kamailio project
by Daniel-Constantin Mierla 16 Dec '15

16 Dec '15

Hello, based on the feedback after the proposal, there were no voices against the new logo. Therefore, as of today, Kamailio project is officially using the new logo -- see it at: - http://www.kamailio.org/w/2015/12/new-logo-for-kamailio-project/ If you display one of the old Kamailio logos, we would encourage to use the new one at your earliest convenience. That will help to propagate it faster. Posting about the change on blogs, forums or social media channels would be appreciated as well. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

RFC: Renaming dialog_ng to ims_dialog
by Daniel-Constantin Mierla 16 Dec '15

16 Dec '15

Hello, I am under the impression that the name dialog_ng creates confusion out there and some people are using it instead of the classic dialog module. Although it was started with goals of reworking dialog module with a different concept (which was discussed mainly by some guys that afterwards changed their job to non-voip area), dialog_ng ended up to be tailored for IMS needs. Probably we should do that refactoring of the dialog module, but meanwhile dialog_ng doesn't refect that and some people are confused by the current naming of the two modules. Practically is more about convenience at this moment and if IMS developers and users think it is not going to be a big overhead for their deployments to be upgraded, I can take care to rename it. So, while general opinion matters, I think we should see first what IMS devs prefer. I am personally not affected that much, so I am fine to keep it like it is now -- in that case, proper notes should be added to documentation, stating that dialog_ng must be used only for IMS (or when the config writer knows very well what she/he is doing). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

4 3

IMC module
by Alexandru Covalschi 16 Dec '15

16 Dec '15

Hello again First of all I wanted to ask if someone ever implemented that http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC Second question is - I don't understand the logic. In description is said: Handles Message method.It detects if the body of the message is a conference command.If so it executes it, otherwise it sends the message to all the members in the room. But why in example (well however it has broken syntax) to IMC manager are sent only messages from chat-rooms? How message from client can possibly reach imc-manager then? Also - when I send message with body "1111111" from user 1001 and imc_manager catches it - I receive 500 command error. Why? :/ All that is working on top of sipjs-demo. -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 3

rtpengine and serial forking with multiple network interfaces
by Alex Balashov 15 Dec '15

15 Dec '15

Hi, It is not very clear to me how to handle serial forking scenarios with rtpengine where there are multiple RTP listeners attached to multiple network interfaces. That is to say, given: rtpengine ... -i net1/xxx.xxx.xxx.xxx -i net2/yyy.yyy.yyy.yyy 1. I attempt to reach SIP gateway 1 via net1, calling rtpengine_manage() in the process: rtpengine_manage("replace-origin replace-session-connection ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)"); 2. The call to gateway 1 fails, and I roll over to gateway 2, reached via net2, with dispatcher. I have some logic that checks how $rd has changed and sets $avp(rtp_dest) accordingly in the dispatcher failure_route[]. 3. In that failure_route[], I call: rtpengine_manage("replace-origin replace-session-connection ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)"); 4. When looking at the SDP for the INVITE on branch #2 going out to gateway 2, it does not appear to utilise media endpoint yyy.yyy.yyy.yyy, but rather continues to represent itself outwardly as xxx.xxx.xxx.xxx, even to gateway #2. I can't figure out what I'm doing wrong here, and the 'rtpengine' documentation only has one mention of serial forking, in the context of the discussion of the 'via-branch' parameter. As far as I can tell, this is not relevant to my use-case; I am not concerned about deleting the whole call from rtpengine, but rather about updating the SDP offer consonantly with serial forking to gateway #2. Any help would be deeply appreciated! Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

Relaying failure codes back to initial server
by Alexandru Covalschi 15 Dec '15

15 Dec '15

Hello everyone! I need to relay 486/408/... other failure codes back to initial INVITE server. Here http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is recommended just to exit failure_route, but that didn't work for me. I need that to let Freeswitch know which cause has ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

3 4

Dispatcher according to called subscriber
by Michael Nielsen 15 Dec '15

15 Dec '15

Hello everyone, I know kamailio can be configured into almsot everything for SIP - and that the dispatcher function is already build for dispatching between multiple sbc's etc... Is it possible to use the dispatcher for routing between multiple sbc's accroding to called subscriber? So I can make it somewhat multi-tentant. So subscriber A always is routed to 1 or more specific sbc's and subscriber B to 1 or more other spcific sbc's?

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