- sr-users - kamailio.org

error running icscf in kamailio 4.4
by Abdelmuhaimen Seaudi 31 Dec '17

31 Dec '17

Hi, After setting up IMS components on kamailio 4.4, i get the following error when IMS client tries to register, what is this error, how can i correct it ? 1(14194) ERROR: <core> [sr_module.c:1766]: get_int_fparam(): Unexpected param type: 1280329042 1(14194) ERROR: ims_icscf [registration.c:78]: I_perform_user_authorization_request(): no int value param passed Abdelmuhaimen Seaudi, CCIE # 25265 Email: aseaudi(a)gmail.com Skype: aseaudi LinkedIn: www.linkedin.com/in/aseaudi Telephone: +2012 84644 733

1 0

Bind IP to Kamailio without restart
by Duarte Rocha 29 Dec '17

29 Dec '17

Greetings, I want to add a new IP to eht0 in my proxy and i want Kamailio to be able to listen to it without me having to restart it. I already have the "listen" parameter set to listen=udp:eth0:5060. Is there any way to reload this parameter it in runtime in order to include the new ip? Is there any alternative to this ? Thanks in advance. Cheers

2 1

PCSCF Access denied issue
by Kranti Kumar 29 Dec '17

29 Dec '17

Hi All, We are using *Kamailio IMS VM* image and we are able to run HSS(/hss.sh) successfully. But while running PCSCF (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg), we are getting below errors. 0(1772) DEBUG: <core> [db.c:209]: db_bind_mod(): using db bind api for db_mysql 0(1772) DEBUG: <core> [db.c:314]: db_do_init2(): connection 0x7f208368d1b8 not found in pool 0(1772) DEBUG: db_mysql [km_my_con.c:98]: db_mysql_new_connection(): opening co nnection: mysql://xxxx:xxxx@server.net1.test/pcscf 0(1772) ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver er ror: Access denied for user 'pcscf'@'pcscf.net1.test' (using password: YES) 0(1772) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(1772) ERROR: ims_usrloc_pcscf [usrloc_db.c:79]: init_db(): unable to co nnect to the database 0(1772) ERROR: ims_usrloc_pcscf [ul_mod.c:203]: mod_init(): Error initializing db connection 0(1772) 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 To fix the above error, we tried below commands *mysql -u root -p GRANT ALL ON *.* TO root IDENTIFIED BY 'root'; quit* - replace bind-address in /etc/mysql/my.cnf to 0.0.0.0 But still we are getting the same issue. Could any one please help us to resolve this issue. Could any one please guide us the troubleshooting steps. Thank you so much in advance. Regards, -kranti

4 6

Trying to fail a call over after 200 OK
by iTalk IP Support Department 28 Dec '17

28 Dec '17

Hi Everyone: I would appreciate your insight regarding a problem we are trying to solve. We are using Kamailio (4.1.8) along with multiple asterisk (13 and above) clusters for various telephony features. We are using the dispatcher module to send calls to the asterisk clusters in round robin and failover algorithms. We need a solution to fail a call and hand it over to another asterisk server after Kamailio has received a 200OK from Asterisk server A. I understand the failover works before the 200OK is sent from asterisk A and calls can easily be sent to asterisk B but again the challenge here to fail the call over to another asterisk server after a 200OK is received from the first one. If this is impossible I was wondering if refer and re-direct is an option to get this done. Thanks in advance for looking at this and for your responses.

2 1

Handling redirect replies and send to new host
by voipspace voipspace 28 Dec '17

28 Dec '17

Hi I am aiming to do the following User calls to Kamailio Switch (A) Switch (A) calls to Kamailio redirect server (B).. (B) runs script to determine route (B) sends Stateless 300 redirect reply to (A). (A) sends ACK to (B). (A) send user call to another host (extracted from header variable from redirect reply) (A) call to (B) rewritehost("x.x.x.x"); t_relay(); (A) On receipt of the 300 response I do not want this redirect to go back to the user. onreply_route { if (t_check_status("3[0-9][0-9]")) { // How do I send ACK to (B) to finish this redirect ? // Then I want a new relay to the host. How would I do it ? } }

3 2

PUBLISH modification.
by Володимир Іванець 27 Dec '17

27 Dec '17

Hello! I am trying to modify PUBLISH packet before handle_publish function execution. The best what I could think of is to make all required changes, update Destination URI to the Kamailio itself and then send it. In this case Kamailio will receive another PUBLISH packet with modified values. I was wondering someone knows cleaner way of achieving this. Thanks!

2 2

How to send "bye" to caller when callee abruptly terminates (Connection reset by peer)
by Pranathi Venkatayogi 26 Dec '17

26 Dec '17

Hi, Currently when callee abruptly closes the connection (TCP RST), Kamailio does not send “BYE” to the caller. Caller is left to believe that he is talking to the callee. What is best way to solve this? I have attempted using tcpops module to track connection RSTs and then send $dlg_bye in event_route[tcp:reset] . In the event_route, I can access only these - $conid , $Ri , $Rp , $si , $sp and $proto variable and do not know how to get to the dlg this connection belongs. Questions: How do I load the dialog associated with this conid? Is there a way to enumerate all dialogs? Thanks in advance, Pranathi.

2 2

Module dispatcher no
by Евгений Голей 26 Dec '17

26 Dec '17

Hello! Sorry for English, I know only Russian . There was a situation when it is necessary to control 1 IP address of the remote gateway from different sockets to kamailio. This is necessary to provide a backup bypass to a remote gateway through two independent Internet providers. Previously, I used the dispatcher module, but in this situation it did not help, I want to understand it the way it should or it's a bug. Here are the settings ------------(PROV-1)----------------\ / \ / (socket:udp:Y1.Y1.Y1.Y1:7060) Remote GW (sip:A1.A1.A1.A1:5060)------ (INTERNET) KAMAILIO \ (socket:udp:X1.X1.X1.X1:7060) \ / -----------(PROV-2)------------------/ kamailio.cfg ... loadmodule "dispatcher.so" ################################################################# modparam("dispatcher", "db_url", "mysql://root:@localhost/kamailio510") modparam("dispatcher", "dst_avp", "$avp(test)") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 2) modparam("dispatcher", "ds_probing_mode", 1) List gw dispatcher ]# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:A1.A1.A1.A1:5060 FLAGS: AP PRIORITY: 2 ATTRS: { BODY: socket=udp:Y1.Y1.Y1.Y1:7060 DUID: MAXLOAD: 0 WEIGHT: 0 RWEIGHT: 0 SOCKET: udp:Y1.Y1.Y1.Y1:7060 } } DEST: { URI: sip:A1.A1.A1.A1:5060 FLAGS: AP PRIORITY: 1 ATTRS: { BODY: socket=udp:X1.X1.X1.X1:7060 DUID: MAXLOAD: 0 WEIGHT: 0 RWEIGHT: 0 SOCKET: udp:X1.X1.X1.X1:7060 } } } } } } Result if the remote gateway is not accessible through the network of the provider 2 (udp:X1.X1.X1.X1:7060), for the unknown reason the status of the first record, which corresponds to the provider 1 (udp:Y1.Y1.Y1.Y1:7060), changes. There are no errors with sockets, OPTIONS packages are sent and a response of 200 OK from the provider is received. 1. OPTIONS requests are sent through provider 2, but there are no answers. But the status changes for some reason at the first. # kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:A1.A1.A1.A1:5060 FLAGS: IP PRIORITY: 2 ATTRS: { BODY: socket=udp:Y1.Y1.Y1.Y1:7060 DUID: MAXLOAD: 0 WEIGHT: 0 RWEIGHT: 0 SOCKET: udp:Y1.Y1.Y1.Y1:7060 } } DEST: { URI: sip:A1.A1.A1.A1:5060 FLAGS: AP PRIORITY: 1 ATTRS: { BODY: socket=udp:X1.X1.X1.X1:7060 DUID: MAXLOAD: 0 WEIGHT: 0 RWEIGHT: 0 SOCKET: udp:X1.X1.X1.X1:7060 } } } } } } At testing it was found out, that камаилио does not allow to supervise a condition at use my circuit. I tried on 5.0.3 and on 5.1.0 the dispatcher works similarly. Please help with advice. Best regards Evgeniy

1 0

Multiple Kamailio Servers + Location Table
by Daniel W. Graham 26 Dec '17

26 Dec '17

When multiple Kamailio servers are using the same database with different server_id's (one active the rest standby in this test), records disappear and are no longer tracked in location database table. Some entries remain, but it never populates to the normal amount unless only one server is using the database. Anyone run into this before? Happened with 5.0 and is happening with 5.1. -dan

4 7

Dispatcher module OPTIONS and event routes: how are they handled by core?
by George Diamantopoulos 21 Dec '17

21 Dec '17

Hello all, I use the dispatcher module extensively for load balancing and fail-over. My kamailio instance is multihomed, and I use the "socket" attribute to determine which socket SIP messages should use for each dispatcher destination, as such: +----+-------+-------------------------+-------+----------+-----------------------------+ | id | setid | destination | flags | priority | attrs | +----+-------+-------------------------+-------+----------+-----------------------------+ | 1 | 0 | sip:192.168.0.1:5060 | 8 | 0 | socket=udp: 10.10.10.1:5060 | | 2 | 1 | sip:111.111.11.1:5060 | 8 | 0 | socket=udp: 44.44.44.1:5060 | | 3 | 1 | sip:222.222.22.2:5060 | 8 | 0 | socket=udp: 55.55.55.1:5060 | +----+-------+-------------------------+-------+----------+-----------------------------+ The dispatcher module uses OPTIONS to probe each destination for availability. When a destination goes down or up, the respective event-route is executed. What I need to do is to be able to "capture" the sending socket used for this probing when a destination becomes unavailable or available in the event-routes. The $fs variable is set, but unfortunately its value does not make sense. Here's an example route and the results that are printed: event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du) $ds $fs $Ru $T_req($fs) $T_req($Ru)\n"); } Now say destination with id = '2' goes down. This is what I get in the logs for the event_route above: ERROR: <script>: Destination down: OPTIONS sip:111.111.11.1:5060 (sip: 192.168.0.1:5060) Contact: <sip:111.111.11.1:5060> udp:10.10.10.1:5060 <null> <null> <null> The xlog PV/log mapping for the above line is the following: $rm: OPTIONS $ru: sip:111.111.11.1:5060 $du: sip:192.168.0.1:5060 $ds: Contact: <sip:111.111.11.1:5060> $Ru: udp:10.10.10.1:5060 The rest are $null. $ru and $ds are consistent with the actual destination being probed, $du and $fs are not (the are set to values corresponding to id = '1', for some reason). This log line is, of course, inaccurate. Not only does it not make sense, but also this is not consistent with messages captured on network interfaces using sngrep: Kamailio does indeed behave as it should, the OPTIONS is sent out to 111.111.11.1 from socket udp:44.44.44.1:5060. But this is not reflected in the log entry above when the event_route is executed. Now the weird part is that this OPTIONS "transaction" (which is locally generated by kamailio) has the $du PV set to the value of another destination (namely the that of id = '1'). As a result, the $fs PV is consistent with that choice for $du. I can verify with sngrep that this is not the case in reality, as the request was sent to destination id = '2' from the correct socket as indicated above. What I would like to ask is whether these "OPTIONS" used by dispatcher for probing go through the request_route processing at some point. This is the only way that would explain the $du PV being set to a false value. If yes, is there any way to prevent this from happening? I need to be able to access the $fs PV when a destination goes up or down, and I can't have any other configuration file routes interfering with that. Thanks! Best regards, George

2 4

strange --dialog in delete state is too old-- log line managing dialog hashes
by David Escartín 21 Dec '17

21 Dec '17

hello all recently we are seeing some weird messages handling with dialogs in Kamailio version 5.0 we sometimes are seeing messages like /usr/local/kamailio/sbin/kamailio[15372]: NOTICE: dialog [dlg_hash.c:249]: dlg_clean_run(): dialog in delete state is too old (0x7fa65445c850 ref 3) /usr/local/kamailio/sbin/kamailio[15372]: NOTICE: dialog [dlg_hash.c:235]: dlg_clean_run(): dialog in early state is too old (0x7fa652d57110 ref 1) we increased the debug description adding some lines to the dialog module code so we could track the calls of the calls that these messages belong to, and we could see that those messages appeared in calls just released at that moment, for example: <134>Nov 4 11:21:38 localhost /usr/local/kamailio/sbin/kamailio[4108]: INFO: mad-localhost-1 Call 97980 / Call-ID 1409565771_82382809(a)195.219.240.46: Creating dialog [8043:21772] with hash id 21772 and hash entry 8043 <134>Nov 4 11:21:38 localhost /usr/local/kamailio/sbin/kamailio[4106]: INFO: mad-localhost-1 Call 97980 / Call-ID 1409565771_82382809(a)195.219.240.46: Status 100, 6610 <134>Nov 4 11:21:39 localhost /usr/local/kamailio/sbin/kamailio[4111]: INFO: mad-localhost-1 Call 97980 / Call-ID 1409565771_82382809(a)195.219.240.46: CANCEL received in A-Leg, relaying downstream <134>Nov 4 11:21:39 localhost /usr/local/kamailio/sbin/kamailio[4112]: INFO: mad-localhost-1 Call 97980 / Call-ID 1409565771_82382809(a)195.219.240.46: Status 487, 6610 <133>Nov 4 11:21:39 localhost /usr/local/kamailio/sbin/kamailio[4139]: NOTICE: dialog [dlg_hash.c:251]: dlg_clean_run(): dialog in delete state is too old (0x7fa0c02a6870 ref 3) with callid '1409565771_82382809(a)195.219.240.46' <129>Nov 4 11:21:39 mad-proxy-inout-1 /usr/local/kamailio/sbin/kamailio[4112]: ALERT: dialog [dlg_handlers.c:1715]: dlg_run_event_route(): after event route - dialog not found [8043:21772] (1/5) (0x7fa0c02a6870) with callid '1409565771_82382809(a)195.219.240.46' we printed the dialog id and entry hash values and we can see there are no other calls creating same values in the previous hours, or using same memory allocation, or same callid, so it seems like there was some kind of strange issue with the dialog timers....¿? By the way, this is happening only few times (80-100 times) a day having many thousands of calls, so it's quite difficult for us to duplicate, we couldn't do it until now. We also tried to use the timer_procs 0 or 1 to use a different proc timer but seems the issue happens in both scenarios. The configuration change we made and seems it was done when these messages started to appear is to use dialog event_route when ended and failed to do some stuff there managing some dialog variables. Does ti make any sense that attempting to use those variables could cause these behaviour? Do you have any idea about it could be or how we can check it deeper? thanks a lot and regards david escartin

2 9

How to add new user to Kamailio using PHP shell_exec?
by Arun NV 21 Dec '17

21 Dec '17

There is a requirement to add a new user to *Kamailio* programmatically using *PHP*. I tried to add username and password to *subscriber* table manually using *SQL* queries, but it won't works because *ha1* and *ha1b* should be added to the table. I don't know how Kamailio generate *ha1* and *ha1b*. Following are the manually added entries using *kamctl* tool. mysql> select * from subscriber limit 2; +----+----------+---------------+--------------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+---------------+--------------+---------------+----------------------------------+----------------------------------+------+ | 9 | 1010101 | 10.101.101.10 | aswwwwwwwsdf | | a37d1785953310c206481ca1a33f16b6 | 7e981130f05a547a738d3c29031e89d0 | NULL | | 10 | 1010102 | 10.101.101.10 | 6eeeeee8a72 | | 6b574f9047206481ca1a33501d7dbdce | cd9a8b89d926f3cb1290311a8cb8a2a8 | NULL | +----+----------+---------------+--------------+---------------+----------------------------------+----------------------------------+------+ Then I tried with *shell_exec* function in *PHP* to use *kamctl* command(*kamctl add username password*). but it will prompt for *Kamailio* read-write user's password. So following command cannot add new users to Kamailio $res = shell_exec("sudo kamctl add new_username new_password); so, please suggest a way to add new user's to *Kamailio* programmatically. <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

3 2

4.4.4 CPU 100%
by Tomas Zanet 21 Dec '17

21 Dec '17

Hi all, I’ve been using Kamailio 4.4.4 for 8 months without any problems but in recent days, I’m having some issues with our server and, when it happens, it cannot get new registrations. Using top I saw a kamailio process is eating all CPU. PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1556 kamailio 20 0 1010884 27624 24520 R 98.5 1.4 150:07.59 kamailio I got backtrace attaching to Kamailio with gdb. Here is the backtrace: #0 0x00007fb996992d6d in accept () at ../sysdeps/unix/syscall-template.S:81 #1 0x00000000005f3615 in handle_new_connect (si=0x7fb9960b4fa8) at tcp_main.c:3894 #2 0x00000000005f7029 in handle_io (fm=0x7fb9961da520, ev=1, idx=-1) at tcp_main.c:4199 #3 0x00000000005c68b2 in io_wait_loop_epoll (h=0xa1fdc0 <io_h>, t=5, repeat=0) at io_wait.h:1065 #4 0x00000000005fa04a in tcp_main_loop () at tcp_main.c:4490 #5 0x00000000004b48ba in main_loop () at main.c:1712 #6 0x00000000004bad12 in main (argc=13, argv=0x7ffd14247c18) at main.c:2631 (gdb) bt At the moment I found the issue, there were 862 sip client registered correctly. Our server is configured to use only TLS and mysql as backend. /usr/local/sbin/kamailio -v version: kamailio 4.4.4 (x86_64/linux) 852c47 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_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: 852c47 compiled on 15:56:18 Dec 15 2016 with gcc 4.8.4 Can anyone address me to find out why this happens? Many thanks, T. Tomas Zanet R&D Software designer tzanet(a)came.com CAME S.p.A.

2 2

Kamailio-IMS VM image issue
by Kranti Kumar 20 Dec '17

20 Dec '17

Hi, I’m trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ <https://clicktime.symantec.com/a/1/vpQmnO-RuJyqB1aW492eOeLkb1WD1yyoCCaN-pca…> My configuration details: HSS - 10.200.8.171 (VM1) PCSCF - 10.200.8.175 (VM2) ICSCF - 10.200.8.178 (VM3) SCSCF - 10.200.8.181 (VM4) While issuing the command ./hss.sh and kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg I am getting errors. Please find the attached log file for the same. "0(1786) ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Access denied for user 'pcscf'@'pcscf.net1.test' (using password: YES) 0(1786) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(1786) ERROR: ims_usrloc_pcscf [usrloc_db.c:79]: init_db(): unable to connect to the database 0(1786) ERROR: ims_usrloc_pcscf [ul_mod.c:203]: mod_init(): Error initializing db connection" I am doubting about my Zone files. Could you please check zone file once. Please find the attached zone files. Thank you so much in advance Regards, -kranti

3 6

PIPELIMIT - limit change after time interval change. bug?
by Juan Priotti 20 Dec '17

20 Dec '17

Hi, I'm making some tests with pipelimit module and when changing the time interval the limit gets changed in a way that it's multiplied by the time interval. For example if I use a time interval of 1 second and a limit of 20 (if (!pl_check("$si", "TAILDROP", "20"))) the module start limiting right after the traffic gets over 20 CPS. But if I change the time interval to 3 seconds, the module starts limiting the traffic right after 60 CPS. This is my configuration: # ----- pipelimit params----- modparam("pipelimit", "timer_interval", 3) modparam("pipelimit", "hash_size", 10) modparam("pipelimit", "db_url", DBURL) modparam("pipelimit", "reply_code", 503) modparam("pipelimit", "reply_reason", "Limiting") ... ... ... request_route { ... ... if(is_method("INVITE")) { $var(limit) = 20; if (!pl_check("$si", "TAILDROP", "$var(limit)")) { xlog("L_INFO", "[$ci] $si - Limiting INVITE using pipe $var(limit) req/sec\n"); pl_drop(); exit; } ..... ..... ..... } } Checking the source for pipelimit module ( https://github.com/kamailio/kamailio/blob/master/src/modules/pipelimit/pipe…) I see that for taildrop it's checking for the limit times the time_interval: case PIPE_ALGO_TAILDROP: ret = (pipe->counter <= pipe->limit * timer_interval) ? 1 : -1; break; I've made a change eliminating the time_interval (set it to 1) from that check and after that I've got the limit working OK, and the traffic gets drop right after it's over the limit (20 cps) using any time_interval. Is there some issue with the timer for the check not being updated by the module settings (timer_interval)? I'm using kamailio 4.4 and I've also tested it in kamailio 4.3 and 5.1 with same results. Thanks in advance, Juan Priotti

2 1

dmq_usrloc, usrloc:db_mode 3, and PostgreSQL
by Alex Balashov 19 Dec '17

19 Dec '17

Hi, Running bleeding-edge branch 5.1 clone here. dmq_usrloc replication works fine when using in-memory only backing for the underlying usrloc module, i.e. db_mode 0. I haven't tried mode 1 and mode 2. I use mode 3, so that contacts are immediately persisted straight to a local database. The database flavour is PostgreSQL in this case. When new contacts replicated via dmq_usrloc come in, I get this in the Kamailio log: Dec 15 19:23:35 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: <core> [db.c:465]: db_use_table(): invalid parameter value Dec 15 19:23:35 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: usrloc [udomain.c:817]: db_load_urecord_by_ruid(): failed to use table location Dec 15 19:23:35 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: WARNING: dmq_usrloc [usrloc_sync.c:147]: delete_contact(): AOR/Contact not found Dec 15 19:23:35 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: <core> [db.c:465]: db_use_table(): invalid parameter value Dec 15 19:23:35 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: usrloc [udomain.c:817]: db_load_urecord_by_ruid(): failed to use table location Dec 15 19:23:35 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: WARNING: dmq_usrloc [usrloc_sync.c:147]: delete_contact(): AOR/Contact not found I added some code to lib/srdb1/db.c to print the parameter name and the location being passed to db_use_table(): */ int db_use_table(db1_con_t* _h, const str* _t) { + LM_ERR("DB handle = %p\n", _h); + if (!_h || !_t || !_t->s) { - LM_ERR("invalid parameter value\n"); + LM_ERR("invalid parameter value %.*s\n", _t->len, _t->s); return -1; } And got this: Dec 15 19:23:36 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: <core> [db.c:462]: db_use_table(): DB handle = (nil) Dec 15 19:23:36 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: <core> [db.c:465]: db_use_table(): invalid parameter value location Dec 15 19:23:36 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: usrloc [udomain.c:631]: db_load_urecord(): failed to use table location Dec 15 19:23:36 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: <core> [db.c:462]: db_use_table(): DB handle = (nil) Dec 15 19:23:36 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: <core> [db.c:465]: db_use_table(): invalid parameter value location Dec 15 19:23:36 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: usrloc [ucontact.c:679]: db_insert_ucontact(): sql use_table failed Dec 15 19:23:36 evaristesys-testbed-2 /usr/local/sbin/kamailio[9395]: ERROR: usrloc [urecord.c:582]: insert_ucontact(): failed to insert in database So, it appears the DB handle being passed here is NULL is in this situation. The same AOR is present in 'ul.dump' RPC command when using db_mode 0. Please advise? :) -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 15

Kamailio v5.0.5 Released
by Daniel-Constantin Mierla 19 Dec '17

19 Dec '17

Hello, Kamailio SIP Server v5.0.5 stable release is out. This is a maintenance release of the previous stable branch, 5.0, that includes fixes since the release of v5.0.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.5 or v5.1.0. For more details about version 5.0.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/12/kamailio-v5-0-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.1, at this moment with its latest release v5.1.0. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Planning Kamailio v5.0.5
by Daniel-Constantin Mierla 19 Dec '17

19 Dec '17

Hello, I am considering to release in the near future a new version from branch 5.0, respectively v5.0.5, a candidate day being next week, on Tuesday, December 19, 2017. If you are aware of any issue in branch 5.0 which is not yet reported on github.com/kamailio/kamailio issues tracker, then do it as soon as possible to have a chance to look at it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 1

Do configuration script file comes under GPL License?
by Seshu Kumar G 18 Dec '17

18 Dec '17

> > Hello Everyone, > We are using only modifying Kamailio configuration files (In IMS application), to tune to our application, without modifying any underlying software. Does configuration script files comes under GPL License? regards GS Kumar

2 1

Fwd: ACK issue TCP in NATed network
by Seshu Kumar 18 Dec '17

18 Dec '17

Hello Everyone, Can someone kindly answer my query below, i am stuck up badly. regards GS Kumar On Wed, Dec 13, 2017 at 12:47 PM, Seshu Kumar <gskumar12345(a)gmail.com> wrote: > Hello Everyone, > > We are using Kamailio IMS (version 5.0.4). We are using Jitsi for making > calls. > Our SIP clients are behind NATed network. > When clients and P-Server is configured to UDP mode, we are able to make > calls sucessfully. > When clients and P-Server is configured to TCP mode, call is not getting > through. > > From logs I found that ACK is not being sent from P-Server to Client. > > On googling i found that this is known problem. > > > Can anyone kindly clarify whether this problem is resolved? > Is there is work around. > Am i not setting any parameter properly? > Let us know if further information is needed. > > Kindly some answer, we are newbies... :) > > Regards > GS Kumar >

2 1

5.0 to 5.1 - No module matching <geoip2> found
by Daniel W. Graham 18 Dec '17

18 Dec '17

Upgraded two installs to 5.1. * Same cfg file from 5.0 * No issues with install 1 Install 2 results in: ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <geoip2> found CRITICAL: <core> [core/cfg.y:3450]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 343, column 71: Can't set module parameter ERROR: <core> [core/sr_module.c:582]: load_module(): could not open module </usr/local/lib64/kamailio/modules/geoip2.so>: libmaxminddb.so.0: cannot open shared object file: No such file or directory CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 122, column 12-22: failed to load module WARNING: <core> [core/ppcfg.c:221]: pp_ifdef_level_check(): different number of preprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 1 ls -l /usr/local/lib64/kamailio/modules/ | grep geo -rwxr-xr-x 1 root root 73800 Dec 14 12:44 geoip2.so If I copy cfg file from install 2 to install 1, no issues with install 1. Any idea what could be going wrong? - dan

3 8

kamailio 4.4.5 Crash
by Andrew Paul 18 Dec '17

18 Dec '17

Hi all, We were using kamailio 4.4.5 and we enabled only for webrtc. We were observing two memory crashed in last two weeks. Oct 10 01:03:03 webrtc /usr/sbin/kamailio[18002]: ERROR: <core> [tcp_read.c:1321]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f21ea8f6198 r: 0x7f21ea8f6218 Oct 10 01:03:04 webrtc /usr/sbin/kamailio[18003]: ERROR: tls [tls_server.c:190]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: shm=9926856 threshold1=9961472 Oct 10 01:03:04 webrtc /usr/sbin/kamailio[18003]: ERROR: <core> [tcp_read.c:1321]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f21ea8f6198 r: 0x7f21ea8f6218 Oct 10 01:03:05 webrtc /usr/sbin/kamailio[18004]: ERROR: tls [tls_server.c:190]: tls_complete_init(): tls: ssl bug #1491 workaround: not enough memory for safe operation: shm=9926856 threshold1=9961472 Oct 10 01:03:05 webrtc /usr/sbin/kamailio[18004]: ERROR: <core> [tcp_read.c:1321]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f21ea8f6198 r: 0x7f21ea8f6218 Is there any known issue for tls module in kamailio_4.4.5 ?. We are using default 64MB for shared memory and 8MB for private memory. Do we need to increase this ?. Regards Andrew

3 6

Kamailio in docker image
by Fatih Yıldız 16 Dec '17

16 Dec '17

Hello everyone, I'd like to know if anyone here installed Kamailio in a docker image. I've found several kamailio images on docker hub but they don't have enough explanation on their description. So I couldn't use them. Can someone help me to build kamailio server or share an image that I can use? I tried tutorials here <https://sonnguyen.ws/install-kamailio-debianubuntu/> and here <https://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/> but something wrong probably and I don't see any kamdbctl file under any installation. Then it stucks and I can't proceed in configuration. Sincerely, *Fatih Yıldız* *Delivery Engineer,* *Phone *: +90 212 285 7575 (ext 117) *Skype* : fatihyldz1 *Team* : dt1(a)defne.com.tr *HelpDesk*: +90 555 996 96 12 This communication is confidential. We only send and receive email on the basis of the terms set out at defne email disclaimer <http://www.defne.com.tr/#%21email-disclaimer/cmjv> Think before you speak. Read before you think. Think before you print. <http://www.defne.com.tr/#!email-disclaimer/cmjv>

3 2

Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)
by Wilkins, Steve 15 Dec '17

15 Dec '17

Hello All, I am looking for a Diagram or such that shows the flow of SIP traffic for a WebRTC Client1 => WebRTC Client2 call using Kamailio in front of Asterisk. I am unable to get Asterisk to find the correct registered clients, which are registered in Kamailio and am hoping verifying the flow will help give me a clue as to what is going on. E.g. Using chrome and tryit-pjsip I have Client1, and Client2 registered in Kamailio. However when I try to connect Client1 to Client2 (make a call), Asterisk has no clue where Client1 and Cleint2 are registered to. Thank you!

2 7

Difference btw Nathelper & Nat Traversal modules
by Melek Oktay 15 Dec '17

15 Dec '17

Hi All, I'd lie to ask the difference between these two modules and get opinion about their use cases. I saw several configuration examples of Kamailio that use RTPProxy. Is it necessary to use RTPProxy? (scenario: public network ---> behind the Kamailio I have FreeSwitch). Clients are behind of NAT & firewall Any thoughts, tips, tricks would be much appreciated. Regards, Angel

2 2

Re: [SR-Users] How to change R-URI Username
by Duarte Rocha 15 Dec '17

15 Dec '17

Thanks for your answer. It doesn't go away when i delete those lines. In fact, i'm using them only to avoid this kind of errors. If i comment out msg_apply_changes() the changes to $rU don't get persistent and i can't find another method to change it.

2 1

How to change R-URI Username
by Duarte Rocha 14 Dec '17

14 Dec '17

Hey everyone. I'm doing some Calling Number and Called Number normalization at the beggining and end of the Kamailio. At the end, i'm doing it in the Dispatch Route. In order to change the numbers, i use uac_replace_to() and uac_replace_from(). However, i don't have a function like this for the R-URI username. I'm trying to change the $rU variable, but i can't apply changes since i get this error from msg_apply_changes() : "cannot apply msg changes after adding record-route header - it breaks conditional 2nd header" Is there any other way to do this operation? Thanks in advance. Cheers

2 1

Unable to enable TLS on Kamailio
by Tomi Hakkarainen 14 Dec '17

14 Dec '17

Hi, I have problem to enable TLS on just installed Kamailio server openSUSE 42.3 (x86_64) VERSION = 42.3 CODENAME = Malachite version: kamailio 5.0.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_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: unknown compiled on 18:06:25 Dec 3 2017 with gcc 4.8.5 I get this on debug log: 0(11336) DEBUG: <core> [core/cfg.y:1642]: yyparse(): loading modules under /usr/lib64/kamailio/modules/ loading modules under config path: /usr/lib64/kamailio/modules/ 0(11336) DEBUG: <core> [core/cfg.y:1623]: yyparse(): loading module tls.so 0(11336) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/tls.so> 0(11336) DEBUG: <core> [core/mem/q_malloc.c:189]: qm_malloc_init(): qm_malloc_init: QM_OPTIMIZE=16384, /ROUNDTO=2048 0(11336) DEBUG: <core> [core/mem/q_malloc.c:191]: qm_malloc_init(): qm_malloc_init: QM_HASH_SIZE=2099, qm_block size=235152 0(11336) DEBUG: <core> [core/mem/q_malloc.c:193]: qm_malloc_init(): qm_malloc_init(0x7f6e001cb000, 67108864), start=0x7f6e001cb000 0(11336) DEBUG: <core> [core/mem/q_malloc.c:202]: qm_malloc_init(): qm_malloc_init: size= 67108864, init_overhead=235256 0(11336) ERROR: tls [tls_init.c:595]: tls_pre_init(): Unable to set the memory allocation functions 0(11336) ERROR: tls [tls_init.c:597]: tls_pre_init(): libssl current mem functions - m: 0x7f6e055b33d0 r: 0x7f6e055b3a30 f: 0x7f6e055b39a0 0(11336) ERROR: tls [tls_init.c:599]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(11336) ERROR: <core> [core/sr_module.c:607]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(11336) CRITICAL: <core> [core/cfg.y:3411]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 150, column 12-19: failed to load module for resolving have compiled openssl from 1.0.2j-fips to openssl version OpenSSL 1.0.2n 7 Dec 2017 Is this information enough to see what we are missing Will provide more info if needed. Any help and suggestions are appreciated. Regards, T

2 8

kamctl rm alice
by Fatih Yıldız 13 Dec '17

13 Dec '17

Hello all, I'm new here and trying to test kamailio. After creating users, I noticed that I made a mistake and create new users. Then I wanted to delete old users but no luck. When I try to delete, it was giving me 2 different errors. I found a way to fix one or skip one, I don't know. It was about CTLENGINE parameter and I uncommented it. :) But second error says that "ERROR: This command requires a database engine - none was loaded" I checked the config files if there is something written like database engine. I found DBENGINE parameter in kamctl file and it was set to MYSQL. Below are the used parameters in kamctl. Could you please check if I'm missing something here? I plan to make some test calls in my local test environment and I installed kamailio 5.0.4 on ubuntu by using Docker. Any help would be appreciated, thanks in advance. SIP_DOMAIN=192.168.99.100 DBENGINE=MYSQL DBHOST=localhost DBNAME=kamailio DBRWUSER="root" DBRPW="password" DBROUSER="root" DBROPW="password" ALIASES_TYPE="db" CTLENGINE="FIFO" OSER_INFO="FIFO" VERBOSE=1 PID_FILE="somewhere" *Fatih Yıldız* *Delivery Engineer,* *Phone *: +90 212 285 7575 (ext 117) *Skype* : fatihyldz1 *Team* : dt1(a)defne.com.tr *HelpDesk*: +90 555 996 96 12 This communication is confidential. We only send and receive email on the basis of the terms set out at defne email disclaimer <http://www.defne.com.tr/#%21email-disclaimer/cmjv> Think before you speak. Read before you think. Think before you print. <http://www.defne.com.tr/#!email-disclaimer/cmjv>

2 5

ACK is TCP in NATed network
by Seshu Kumar 13 Dec '17

13 Dec '17

Hello Everyone, We are using Kamailio IMS (version 5.0.4). We are using Jitsi for making calls. Our SIP clients are behind NATed network. When clients and P-Server is configured to UDP mode, we are able to make calls sucessfully. When clients and P-Server is configured to TCP mode, call is not getting through. >From logs I found that ACK is not being sent from P-Server to Client. On googling i found that this is known problem. Can anyone kindly clarify whether this problem is resolved? Is there is work around. Am i not setting any parameter properly? Let us know if further information is needed. Kindly some answer, we are newbies... :) Regards GS Kumar

1 0

Re: [SR-Users] Exec - call python script with arguments
by otron2016＠gmail.com 13 Dec '17

13 Dec '17

Probably not the problem though it could be. ...try: "python /usr/local/etc/kamailio/script.py arg1 arg2 arg3" instead--kamailio doesn't have your shell env and might not pass the script to python even if the script has the #! declaration. Also, if you pasted the exact command I think there's no space between .py and arg1 (or that could just be my phone). Hth. Sent from Samsung Mobile -------- Original message -------- From: "Daniel W. Graham" <dan(a)cmsinter.net> Date: 12/10/2017 1:04 PM (GMT-06:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Exec - call python script with arguments Trying to call a python script using exec_msg and haven’t had any luck. The following works: exec_msg("echo ‘TEST’ > /usr/local/etc/kamailio/test.txt"); The following works from shell: /usr/local/etc/kamailio/script.py arg1 arg2 arg3 The following does not work from kamailio: exec_msg("/usr/local/etc/kamailio/script.py arg1 arg2 arg3"); (The arguments are just strings for test purposes) Kamailio is running as kamailio user Permissions: 755 script.py Any idea what could be going wrong? -dan

2 1

blf with kamailio not sending NOTIFY
by Ravi Kuru 12 Dec '17

12 Dec '17

Hi, I set up BLF but not working, phone sending SUBSCRIBE message then kamailio is sending 202 OK but not sending NOTIFY message. What could be the issue. Last login: Mon Dec 11 10:56:53 on ttys000 raveechdransmbp:~ rkuru$ ssh -p44422 kuru.ravi(a)67.227.65.10 kuru.ravi(a)67.227.65.10's password: Last login: Sun Dec 10 10:25:45 2017 from 216.123.66.49 [kuru.ravi@rpx1 ~]$ su - Password: Last login: Sun Dec 10 10:25:53 EST 2017 on pts/0 [root@rpx1 ~]# cd /usr/local/etc/kamailio/ [root@rpx1 kamailio]# vi kamailio.cfg [root@rpx1 kamailio]# vi [root@rpx1 kamailio]# cd [root@rpx1 ~]# vi a Request-Line: SUBSCRIBE sip:8665336444_202@67.227.65.10 SIP/2.0 Method: SUBSCRIBE Request-URI: sip:8665336444_202@67.227.65.10 Request-URI User Part: 8665336444_202 Request-URI Host Part: 67.227.65.10 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 66.46.51.154:21389;branch=z9hG4bK1022076515;rport Transport: UDP Sent-by Address: 66.46.51.154 Sent-by port: 21389 Branch: z9hG4bK1022076515 RPort: rport From: <sip:8665336444_201@67.227.65.10>;tag=1691126593 SIP from address: sip:8665336444_201@67.227.65.10 SIP from address User Part: 8665336444_201 SIP from address Host Part: 67.227.65.10 SIP from tag: 1691126593 To: <sip:8665336444_202@67.227.65.10> SIP to address: sip:8665336444_202@67.227.65.10 SIP to address User Part: 8665336444_202 SIP to address Host Part: 67.227.65.10 Call-ID: 1930543912-5060-5(a)BA.B.B.BEC CSeq: 20031 SUBSCRIBE Sequence Number: 20031 Method: SUBSCRIBE Contact: <sip:8665336444_201@66.46.51.154:21389> Contact URI: sip:8665336444_201@66.46.51.154:21389 Contact URI User Part: 8665336444_201 Contact URI Host Part: 66.46.51.154 Contact URI Host Port: 21389 Proxy-Authorization: Digest username="8665336444_201", realm="67.227.65.10", nonce="Wil7QVopehWJO/uGzcWNmD3vlFL39ztq", uri=" sip:8665336444_202@67.227.65.10", response="c7bf71e9f7f4d9720655531cbd768cff", algorithm=MD5 Authentication Scheme: Digest Username: "8665336444_201" Realm: "67.227.65.10" Nonce Value: "Wil7QVopehWJO/uGzcWNmD3vlFL39ztq" Authentication URI: "sip:8665336444_202@67.227.65.10" Digest Authentication Response: "c7bf71e9f7f4d9720655531cbd768cff" Algorithm: MD5 X-Grandstream-PBX: true [Expert Info (Note/Undecoded): Unrecognised SIP header (X-Grandstream-PBX)] [Message: Unrecognised SIP header (X-Grandstream-PBX)] [Severity level: Note] [Group: Undecoded] Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.5.29 Expires: 1800 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Session Initiation Protocol (202) Status-Line: SIP/2.0 202 OK Status-Code: 202 [Resent Packet: False] [Request Frame: 295] [Response Time (ms): 0] Message Header Via: SIP/2.0/UDP 66.46.51.154:21389 ;branch=z9hG4bK1022076515;rport=21389;received=66.46.51.154 Transport: UDP Sent-by Address: 66.46.51.154 Sent-by port: 21389 Branch: z9hG4bK1022076515 RPort: 21389 Received: 66.46.51.154 From: <sip:8665336444_201@67.227.65.10>;tag=1691126593 SIP from address: sip:8665336444_201@67.227.65.10 SIP from address User Part: 8665336444_201 SIP from address Host Part: 67.227.65.10 SIP from tag: 1691126593 To: <sip:8665336444_202@67.227.65.10 >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-fbce SIP to address: sip:8665336444_202@67.227.65.10 SIP to address User Part: 8665336444_202 SIP to address Host Part: 67.227.65.10 SIP to tag: a6a1c5f60faecf035a1ae5b6e96e979a-fbce Call-ID: 1930543912-5060-5(a)BA.B.B.BEC CSeq: 20031 SUBSCRIBE Sequence Number: 20031 Method: SUBSCRIBE Expires: 1800 Contact: <sip:67.227.65.19:5060> Contact URI: sip:67.227.65.19:5060 Contact URI Host Part: 67.227.65.19 Contact URI Host Port: 5060 Server: kamailio (5.0.3 (x86_64/linux)) Content-Length: 0 this is the my kamailio.cfg Last login: Mon Dec 11 10:56:53 on ttys000 raveechdransmbp:~ rkuru$ ssh -p44422 kuru.ravi(a)67.227.65.10 kuru.ravi(a)67.227.65.10's password: Last login: Sun Dec 10 10:25:45 2017 from 216.123.66.49 [kuru.ravi@rpx1 ~]$ su - Password: Last login: Sun Dec 10 10:25:53 EST 2017 on pts/0 [root@rpx1 ~]# cd /usr/local/etc/kamailio/ [root@rpx1 kamailio]# vi kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_PSTN #!define WITH_NAT #!define WITH_ANTIFLOOD #!define WITH_PRESENCE # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB "kamailio.cfg" 1232L, 34401C Last login: Mon Dec 11 10:56:53 on ttys000 raveechdransmbp:~ rkuru$ ssh -p44422 kuru.ravi(a)67.227.65.10 kuru.ravi(a)67.227.65.10's password: Last login: Sun Dec 10 10:25:45 2017 from 216.123.66.49 [kuru.ravi@rpx1 ~]$ su - Password: Last login: Sun Dec 10 10:25:53 EST 2017 on pts/0 [root@rpx1 ~]# cd /usr/local/etc/kamailio/ [root@rpx1 kamailio]# vi kamailio.cfg # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #mhomed=1 #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "unixodbc://telcan:t3lc4n@localhost/MySQLkamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "unixodbc://test:test@localhost/MySQLasterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_DLG 9 #!define FLT_DLGINFO 10 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="rp-na1.sipconfig.net" alias="na-rpx1.sipconfig.net" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:38.64.154.234:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: #pstn.gw_ip = "67.227.65.2" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "67.227.65.10" desc "PSTN GW Address" #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "67.227.65.10" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "67.227.65.10" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "db_unixodbc.so" #loadmodule "mi_fifo.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" #loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "sqlops.so" loadmodule "dialog.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ modparam("registrar", "append_branches", 1) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 4) modparam("registrar", "retry_after", 60) # max value for expires of registrations #modparam("registrar", "min_expires", 0) modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- registrar params ----- #modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ #modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) #modparam("registrar", "retry_after", 30) # max value for expires of registrations #modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU #modparam("registrar", "gruu_enabled", 1) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") ##!ifdef WITH_ASTERISK #modparam("auth_db", "user_column", "name") #modparam("auth_db", "password_column", "sippasswd") #modparam("auth_db", "db_url", DBASTURL) #modparam("auth_db", "version_table", 0) ##!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) ##!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:67.227.65.10:5060" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 2) modparam("presence", "fetch_rows", 1000) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) # ----- presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_single_dialog", 0) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) #!endif # ----- dialog params ----- modparam("dialog", "db_url", DBURL) modparam("dialog", "enable_stats", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 3) modparam("pike", "reqs_density_per_unit", 1000) modparam("pike", "remove_latency", 10) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif # ------- Load-balancer params ------ modparam("dispatcher", "db_url","unixodbc://test:test@localhost /MySQLkamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@67.227.65.10") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); xlog("L_INFO","AFTER LOCATION to ROUTE DEVICE ------- : $ci"); ####route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); dlg_manage(); } xlog("L_INFO","----- This is to device < $du > asterisk server -------"); if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); exit; } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$"){ if(sd_lookup("speed_dial")){ route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("REFER")){ xlog("L_INFO"," WITH_ASTERISK WITHINDLG SIP REFER METHOD -------"); } #!endif xlog("L_INFO"," FROM ASTERISK TO DEVICE -------"); $avp(oexten) = $rU; if (!lookup("location")) { xlog("L_INFO","LOOK UP LOCATION FAILED -------"); $var(rc) = $rc; xlog("L_INFO"," *********** $var(rc) *****************"); #route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } xlog("L_INFO","LOCATION FROM ASTERISK TO DEVICE -------"); # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ##!ifdef WITH_ASTERISK # if (!auth_check("$fd", "sipusers", "1")) { ##!else if (!auth_check("$fd", "subscriber", "1")) { ##!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("L_INFO","SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { #route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk # Send to Asterisk route[TOASTERISK] { # only local users allowed to call if(is_method("REFER")){ xlog("L_INFO"," SIP REFER METHOD -------"); } if(from_uri!=myself) { xlog("L_INFO","[$fU@$si:$sp]{$rm} *************** From Asterisk Box to Device$du\n"); }else{ if(is_method("INVITE")){ xlog("L_INFO"," device to Asterisk Box $du ++++\n"); route(RELAY); exit; } } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } ds_mark_dst("P"); $var(x) = $au; $var(x) = $(var(x){s.substr,0,3}); if(!ds_select_dst($(var(x){s.int}), "1")) { $var(rip) = $sel(cfg_get.asterisk.bindip); }else{ $var(rip1) = $du; $var(rip) = $(var(rip1){s.replace,:5060,}) ; $var(rip) = $(var(rip){s.replace,sip:,}) ; xlog("L_INFO","$au <------- Loadbalancer REGISTER to Asterisk $var(rip) ------->"); } if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 2

newbie question - routing sip-calls
by thomas peterseil 12 Dec '17

12 Dec '17

hello, sorry for this very basic question, but i am a new kamailio user. i installed kamailio and calls between users work fine. now i would like to make it possible to route outbound sip calls. so when a user dials for example 21 the call should be routed to wbdemo(a)zipdx.com thank you so much for your help! best regards, thomas peterseil

2 1

Re: [SR-Users] Unable to enable TLS on Kamailio
by otron2016＠gmail.com 12 Dec '17

12 Dec '17

Broken is in the eyes of the beholder: well designed cryptographic code wants to ensure that information (keys, cleartext) doesn't leak via unsanitized memory (there are many ways, both within and beyond calling programs); the easy and more foolproof way to do that for the cryptography programmer is often to use a memory manager that takes care of that, such as jemalloc (with appropriate configuration parameters). If you make security representations (and the certificate is reasonably construed to make a security representation) you shouldn't bypass this unless you verify that you prevent all possible information leaks. From armslength, you might just try to use jemalloc as kamailio's mm library, but even there it would be necessary to be really careful about kamailio freeing sensitive memory immediately after use--everywhere that happens. That's why it's probably easier to just let a properly implemented crypto library do what it's designed to do. Sent from Samsung Mobile -------- Original message -------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: 12/12/2017 2:26 AM (GMT-06:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>,Tomi Hakkarainen <tpaivaa(a)gmail.com> Subject: Re: [SR-Users] Unable to enable TLS on Kamailio Hello, there were some broken versions of openssl that didn't allow anymore to set custom memory manager. The only option is to upgrade libssl to a version that doesn't expose the issue. If you search on kamailio issues tracker on gihub.com, there should be one closed about this topic. Cheers, Daniel On 11.12.17 22:20, Tomi Hakkarainen wrote: Hi, I have problem to enable TLS on just installed Kamailio server openSUSE 42.3 (x86_64) VERSION = 42.3 CODENAME = Malachite version: kamailio 5.0.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_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: unknown compiled on 18:06:25 Dec 3 2017 with gcc 4.8.5 I get this on debug log: 0(11336) DEBUG: <core> [core/cfg.y:1642]: yyparse(): loading modules under /usr/lib64/kamailio/modules/ loading modules under config path: /usr/lib64/kamailio/modules/ 0(11336) DEBUG: <core> [core/cfg.y:1623]: yyparse(): loading module tls.so 0(11336) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/tls.so> 0(11336) DEBUG: <core> [core/mem/q_malloc.c:189]: qm_malloc_init(): qm_malloc_init: QM_OPTIMIZE=16384, /ROUNDTO=2048 0(11336) DEBUG: <core> [core/mem/q_malloc.c:191]: qm_malloc_init(): qm_malloc_init: QM_HASH_SIZE=2099, qm_block size=235152 0(11336) DEBUG: <core> [core/mem/q_malloc.c:193]: qm_malloc_init(): qm_malloc_init(0x7f6e001cb000, 67108864), start=0x7f6e001cb000 0(11336) DEBUG: <core> [core/mem/q_malloc.c:202]: qm_malloc_init(): qm_malloc_init: size= 67108864, init_overhead=235256 0(11336) ERROR: tls [tls_init.c:595]: tls_pre_init(): Unable to set the memory allocation functions 0(11336) ERROR: tls [tls_init.c:597]: tls_pre_init(): libssl current mem functions - m: 0x7f6e055b33d0 r: 0x7f6e055b3a30 f: 0x7f6e055b39a0 0(11336) ERROR: tls [tls_init.c:599]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(11336) ERROR: <core> [core/sr_module.c:607]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(11336) CRITICAL: <core> [core/cfg.y:3411]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 150, column 12-19: failed to load module for resolving have compiled openssl from 1.0.2j-fips to openssl version OpenSSL 1.0.2n 7 Dec 2017 Is this information enough to see what we are missing Will provide more info if needed. Any help and suggestions are appreciated. Regards, T _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

2 1

Dispatcher flow regarding weight and priority.
by Joel Serrano 12 Dec '17

12 Dec '17

Hi all, I have a doubt regarding how dispatcher selects a node... >From the docs: *[...]* *alg* - the algorithm used to select the destination address. The parameter can be an integer or a variable holding an interger. - “4” - round-robin (next destination). - “8” - select destination sorted by priority attribute value (serial forking ordered by priority). - “9” - use weight based load distribution. You have to set the attribute 'weight' per each address in destination set. *[...]* I normally use "4", so dispatcher round-robins through the destinations of a setid, but I have now the following doubt: If a trunk has 3 nodes, say 1.1.1.1, 2.2.2.2, 3.3.3.3 Can I do: Send all calls to 1.1.1.1, if it goes down, then do a 80%-20% between 2.2.2.2 and 3.3.3.3. if the trunk has a correct DNS SRV setup with weight and priority it's easy to achieve, but in this case there is no SRV or nothing, just 3 IPs. So can this logic be achieved with dispatcher directly? Can do some combination of alg "8" and alg "9"? Or would the correct approach need two separate setid groups and handle that logic in failure route? Thanks, Joel.

2 2

pua.publish using RPC
by Alex Balashov 12 Dec '17

12 Dec '17

Hello, Is pua.publish strictly an MI function, or is it possible nowadays to call it via the RPC channel, and specifically, using jsonrpc_exec()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

Re: [SR-Users] Kamailio does not add port number to addresses in Via, Record-Route headers
by George Diamantopoulos 12 Dec '17

12 Dec '17

Hello again, Indeed this issue does not manifest at all. I'm awfully sorry for the false alarm, and on release day no less! The problem was there was a lingering DNAT rule in iptables, which would translate port 5066 to port 5060. The deployment script injected this as it was carried over from our legacy platform. Of course, I kept banging my head against the wall here because sngrep wouldn't show the DNAT's effect as it captures traffic from the NIC directly: it would show a REGISTER arriving on 5066, but the dport was masqueraded before being handed over to kamailio. Similarly for the outgoing INVITE. NAT is wrong in so many ways... :-) BR, George On 11 December 2017 at 18:17, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > I did a quick test and all looks fine, ports are set in via and > record-route, in my config I have: > > record_route(); > > $fs="udp:127.0.0.1:5080"; > $du = "sip:127.0.0.1:9"; > t_relay(); > exit; > > Then sending an OPTIONS resulted in the trace shown below. > > Cheers, > Daniel > > U 2017/12/11 17:14:47.108430 127.0.0.1:56729 -> 127.0.0.1:5060 > OPTIONS sip:test@127.0.0.1 SIP/2.0. > Via: SIP/2.0/UDP 192.168.178.84:62516;branch=z9hG4bK.3aaddf68;rport;alias. > From: sip:sipsak@192.168.178.84:62516;tag=16d1c24. > To: sip:test@127.0.0.1. > Call-ID: 23927844(a)192.168.178.84. > CSeq: 1 OPTIONS. > Contact: sip:sipsak@192.168.178.84:62516. > Content-Length: 0. > Max-Forwards: 70. > User-Agent: sipsak 0.9.7pre. > Accept: text/plain. > . > > > U 2017/12/11 17:14:51.010251 127.0.0.1:5080 -> 127.0.0.1:9 > OPTIONS sip:test@127.0.0.1 SIP/2.0. > Record-Route: <sip:127.0.0.1:5080;r2=on;lr>. > Record-Route: <sip:127.0.0.1;r2=on;lr>. > Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK61bd. > b2882fea15c488761489f8ef588efbe1.0. > Via: SIP/2.0/UDP 192.168.178.84:62516;received=127.0.0.1;branch=z9hG4bK. > 3aaddf68;rport=56729;alias. > From: sip:sipsak@192.168.178.84:62516;tag=16d1c24. > To: sip:test@127.0.0.1. > Call-ID: 23927844(a)192.168.178.84. > CSeq: 1 OPTIONS. > Contact: sip:sipsak@192.168.178.84:62516. > Content-Length: 0. > Max-Forwards: 69. > User-Agent: sipsak 0.9.7pre. > Accept: text/plain. > . > > On 11.12.17 16:37, George Diamantopoulos wrote: > > Hello all, > > I have the following issue in my configuration, tested with 5.2.0-rc1 so > far: > > At some point, I set the $fs pseudovariable to force a request to be > relayed through a specific socket. Although this is honoured by kamailio > (i.e. the request does indeed leave the kamailio host from the respective > socket), the port number is not added to the Via and RR headers. As a > result, all replies to the request, as well as all subsequent requests from > the other SIP UA are relayed to the default port, 5060. Here's an example: > > SIP UAC to kamailio: > INVITE 192.168.1.1:5060 ---> 192.168.1.254:5060 > Kamailio to UAS ($fs is set): > INVITE 2.2.2.2:5066 ---> 3.3.3.3:5060 > Topmost Via in request relayed by kamailio is: > SIP/2.0/UDP 2.2.2.2;branch=aaaaaaaaaaaaaa <- port 5066 is not added > Topmost RR in request relayed by kamailio is: > <sip:2.2.2.2;r2=on;lr;did=bbbbbbb;nat=yes> <- port 5066 is not added > RESULT: Reply from UAS is sent to 2.2.2.2:5060 > > Is this behaviour valid? Am I missing anything? Kamailio is configured to > listen on both sockets on IP 2.2.2.2, namely: a) udp:2.2.2.2:5060 and b) > 2.2.2.2:5066. Thanks. > > BR, > George > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com > >

2 1

Change Calling Number
by Duarte Rocha 11 Dec '17

11 Dec '17

Hey I'm trying to do some normalization on my Calling Number after Kamailio chooses the Gateway in which it will deliver the request. So, i'm trying to do the operation in the DISPATCH route. However, in this route, msg_apply_changes() doesn't work and returns (-1). I've tried to do this on a Branch route but Kamailio doesn't allow me to use msg_apply_changes() there and otherwise changes won't be persistent. Is there a way to modify $fU at this stage? Cheers

2 1

Kamailio v5.1.0 Released - new major version is out
by Daniel-Constantin Mierla 11 Dec '17

11 Dec '17

Kamailio v5.1.0 is out – it comes with 9 new modules and a significant set of improvements touching more than 90 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all developers and community members that made possible this release. A consistent effort was directed to export most of the config functions to the KEMI framework, allowing to write SIP routing logic in other scripting languages executed by the embedded interpreters for Lua, JavaScript, Python or Squirrel, with the option to reload the routing script without restarting Kamailio. Enjoy Kamailio v5.1.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2018 - http://www.kamailioworld.com

1 0

Kamailio does not add port number to addresses in Via, Record-Route headers
by George Diamantopoulos 11 Dec '17

11 Dec '17

Hello all, I have the following issue in my configuration, tested with 5.2.0-rc1 so far: At some point, I set the $fs pseudovariable to force a request to be relayed through a specific socket. Although this is honoured by kamailio (i.e. the request does indeed leave the kamailio host from the respective socket), the port number is not added to the Via and RR headers. As a result, all replies to the request, as well as all subsequent requests from the other SIP UA are relayed to the default port, 5060. Here's an example: SIP UAC to kamailio: INVITE 192.168.1.1:5060 ---> 192.168.1.254:5060 Kamailio to UAS ($fs is set): INVITE 2.2.2.2:5066 ---> 3.3.3.3:5060 Topmost Via in request relayed by kamailio is: SIP/2.0/UDP 2.2.2.2;branch=aaaaaaaaaaaaaa <- port 5066 is not added Topmost RR in request relayed by kamailio is: <sip:2.2.2.2;r2=on;lr;did=bbbbbbb;nat=yes> <- port 5066 is not added RESULT: Reply from UAS is sent to 2.2.2.2:5060 Is this behaviour valid? Am I missing anything? Kamailio is configured to listen on both sockets on IP 2.2.2.2, namely: a) udp:2.2.2.2:5060 and b) 2.2.2.2:5066. Thanks. BR, George

2 1

Re: [SR-Users] Failed to find command http_client_query
by Alexandru Covalschi 11 Dec '17

11 Dec '17

Hello, Wanted to ask if you found something ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493 > 24 нояб. 2017 г., в 22:19, Alexandru Covalschi <568691(a)gmail.com> написал(а): > > Hello, sharing whole config except vars.cfg file, where values like API_PROTO, DBURL etc. are defined. > > <kamailio_504.cfg> > > > ________________________________ > Alexandru Covalschi > VoIP Engineer and System Administrator > tel: +373 673 98 493 > >> 24 нояб. 2017 г., в 13:21, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >> >> Hello, >> >> can you provide here all the http_client_query(...) you have in your config and list in which route block are they used (like request_route, failure_route, ...)? You can replace IP addresses, usernames, passwords with other strings, but if you use any kamailio variables (e.g. $rU), let them in place. >> Cheers, >> Daniel >> >> On 23.11.17 20:54, Alexandru Covalschi wrote: >>> sip:172.21.0.14:5060 <sip:172.21.0.14:5060> is the local address of the server, it seems like a bug in http_client, as I don't have such http_query call anywhere :/ >>> just re-checked >>> Also the fact that it works perfectly on 4.4.6 tells that it's not a configuration issue >>> >>> ________________________________ >>> Alexandru Covalschi >>> VoIP Engineer and System Administrator >>> tel: +373 673 98 493 >>> >>>> 23 нояб. 2017 г., в 16:59, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >>>> >>>> There was a mistake in the exports structure, the name of the function with 4 parameter was http_query instead of http_client_query. I pushed a patch to fix it. >>>> Then, related to the issue with cannot resolve the host, the debug messages show an invalid http url being used: >>>> >>>> Nov 21 04:18:20 ast1 /usr/sbin/kamailio[20804]: DEBUG: http_client [functions.c:146]: curL_query_url(): ****** ##### CURL URL [sip:172.21.0.14:5060 <sip:172.21.0.14:5060>] >>>> >>>> The it fails to do the dns stuff with it... >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 22.11.17 22:38, Alexandru Covalschi wrote: >>>>> Just wanted to ask if you need my help to test/check/etc. >>>>> >>>>> ________________________________ >>>>> Alexandru Covalschi >>>>> VoIP Engineer and System Administrator >>>>> tel: +373 673 98 493 >>>>> >>>>>> 21 нояб. 2017 г., в 11:58, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>> >>>>>> Kamailio is actually starting, please take a look at that debug=4 log file: https://filebin.net/r54y3jkmzdyk9zak/syslog_http_client_query_failed.txt <https://filebin.net/r54y3jkmzdyk9zak/syslog_http_client_query_failed.txt> >>>>>> at line 1042 http_client module is successfully loading, but at line 12785 it fails to load http_client_query >>>>>> >>>>>> Also please check line 50385 in https://filebin.net/352vff41fwx7uowg/syslog_http_query_curl_6.txt <https://filebin.net/352vff41fwx7uowg/syslog_http_query_curl_6.txt> >>>>>> I'm using http_query here with 5.0.4 http_client, you can see that http_client makes a request, but afterwards it returns curl error 6, but the weirdest thing is - at line 19887 I request token from the very same IP address, only url is changed, and it actually works fine. >>>>>> ________________________________ >>>>>> Alexandru Covalschi >>>>>> VoIP Engineer and System Administrator >>>>>> tel: +373 673 98 493 >>>>>> >>>>>>> 21 нояб. 2017 г., в 10:44, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >>>>>>> >>>>>>> But then kamailio is not starting, or? >>>>>>> >>>>>>> Can you provide all the error message as they are printed in syslog file? File, line and other details are part of the logs message, allowing to identify the location in the source code where that happens ... >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> On 21.11.17 09:38, Alexandru Covalschi wrote: >>>>>>>> It is still one of the issues >>>>>>>> >>>>>>>> ________________________________ >>>>>>>> Alexandru Covalschi >>>>>>>> VoIP Engineer and System Administrator >>>>>>>> tel: +373 673 98 493 >>>>>>>> >>>>>>>>> 21 нояб. 2017 г., в 9:40, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >>>>>>>>> >>>>>>>>> Hello, >>>>>>>>> >>>>>>>>> in your email in this thread, you said: >>>>>>>>> >>>>>>>>> > When I try to change http_query to http_client_query it fails to load with error "failed to find command http_client_query". >>>>>>>>> >>>>>>>>> Is it still the case, or that was sorted out? >>>>>>>>> >>>>>>>>> Cheers, >>>>>>>>> Daniel >>>>>>>>> >>>>>>>>> On 20.11.17 22:43, Alexandru Covalschi wrote: >>>>>>>>>> Works ok on 4.4.6. So I suppose it's a weird 5.0.4 + my environment bug. I'm rolling back to 4.4.6 right now, will keep one 5.0.4 machine for testing (on debian 9) >>>>>>>>>> Let me know if I can provide any dumps/cores etc. >>>>>>>>>> >>>>>>>>>> ________________________________ >>>>>>>>>> Alexandru Covalschi >>>>>>>>>> VoIP Engineer and System Administrator >>>>>>>>>> tel: +373 673 98 493 >>>>>>>>>> >>>>>>>>>>> 20 нояб. 2017 г., в 23:29, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>>>>>>> >>>>>>>>>>> So the weirdest thing is Kamailio actually sends the request, but it returns curl error 6 in the answer variable. I have a plenty of clonned servers and same behaviour is on all of them, also just now tried on a fresh debian 9 - same result. I'm inside a virtual machine (vmware), but not sure it matters... >>>>>>>>>>> >>>>>>>>>>> ________________________________ >>>>>>>>>>> Alexandru Covalschi >>>>>>>>>>> VoIP Engineer and System Administrator >>>>>>>>>>> tel: +373 673 98 493 >>>>>>>>>>> >>>>>>>>>>>> 20 нояб. 2017 г., в 22:37, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>>>>>>>> >>>>>>>>>>>> I often get WARNING: http_client [functions.c:281]: curL_query_url(): Couldn't resolve host, while I don't even use hostnames. >>>>>>>>>>>> After I changed IP address to hostname (which I've added into /etc/hosts) nothing changed. It seems like a weird bug, but I've no idea - is it CURL or kamailio, but when I execute curl from shell it is working fine. I have google dns in resolve.conf and nslookup is also working ok. >>>>>>>>>>>> Can anyone help me with that? >>>>>>>>>>>> >>>>>>>>>>>> ________________________________ >>>>>>>>>>>> Alexandru Covalschi >>>>>>>>>>>> VoIP Engineer and System Administrator >>>>>>>>>>>> tel: +373 673 98 493 >>>>>>>>>>>> >>>>>>>>>>>>> 20 нояб. 2017 г., в 22:06, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>>>>>>>>> >>>>>>>>>>>>> t >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> >>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Daniel-Constantin Mierla >>>>>>>>> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >>>>>>>>> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >>>>>>>>> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierla >>>>>>> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >>>>>>> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >>>>>>> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/> >>>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla >>>> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >>>> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >>>> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/> >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/>

2 2

can kamailio send 183 to caller
by 赵国杰 11 Dec '17

11 Dec '17

Hello guys, Is it possible for kamailio to initiate a 183 to caller? Thanks

3 6

Releasing Kamailio v5.1.0
by Daniel-Constantin Mierla 11 Dec '17

11 Dec '17

Hello, with no relevant new bugs reported lately, it is time to fix the date for releasing next major version, respectively Kamailio v5.1.0. I am proposing Monday, December 11, 2018, as the release date for v5.1.0, with a fallback option to next day, if a bit of time is still needed. Currently there are ongoing efforts to make rpm packaging a first class citizen via docker and obs, credits to Sergey Safarov for it, and get (auto-generated) reference list of Kemi functions, credits to Samuel Förnes for it. In the next two weeks, they can get in good shape and also do a bit more testing as well as update relevant tutorials. Regarding Kemi exports, if you are using a module, check the master branch version to see if it has any sr_kemi_t exports, if not, report it in the case you want to try using the new config framework that allows reloading routing logic script without restart (you will have to use Lua, JavaScript or Squirrel scripting). And of course, if you are aware of any problem not yet reported to the bug tracker, open an issue as soon as possible to allow time to try to fix it before the release: * https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 2

Exec - call python script with arguments
by Daniel W. Graham 10 Dec '17

10 Dec '17

Trying to call a python script using exec_msg and haven't had any luck. The following works: exec_msg("echo 'TEST' > /usr/local/etc/kamailio/test.txt"); The following works from shell: /usr/local/etc/kamailio/script.py arg1 arg2 arg3 The following does not work from kamailio: exec_msg("/usr/local/etc/kamailio/script.py arg1 arg2 arg3"); (The arguments are just strings for test purposes) Kamailio is running as kamailio user Permissions: 755 script.py Any idea what could be going wrong? -dan

2 2

TOPOS - record timeouts
by Daniel W. Graham 10 Dec '17

10 Dec '17

Looking for clarification on how the timers work, such as 'dialog_expire'. Are these hard-set timeouts meaning if a dialog lasts longer than 'dialog_expire' there could be risk of the dialog not completing successfully? If yes, any plans to integrate with DIALOG module to keep topos records up-to-date per active dialogs? Thanks, -dan

2 2

Need help troubleshooting messages that never made it to the client
by Andrew Chen 09 Dec '17

09 Dec '17

Hi all, So for a while now I've been working with our engineering team to troubleshoot these random events where the UPDATE message sent from the server never made it to the client side. Here is the the topology: Server -(tcp)-> kamailio -(tls)-> client So looks pretty straight forward and I have logging messages stating t_relay() was successful in sending the message. The question is how do we know t_relay really successfully sent it? Does it actually monitor the physical interface as it goes out? Please advise. Thanks. -- Andy Chen -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 6

cfg_get in async_worker causes segfault (v4.4)
by Malcolm O'Hare 09 Dec '17

09 Dec '17

I want to experiment with using the async module. I added the module and updated my request_route to call async_task_route() and put all the existing routing logic in a new route. Unfortunately my kamailio is crashing looking up a cfg_get variable. The exact same route logic works fine in the normal child process, its only when its run in a async_worker that it fails. Is this a known issue? I can see in the core file that there is no group handle in the cfg_group_t (its 0x0), but I know nothing about how that gets set and where. Thanks in advance for any help Malcolm

2 1

Forwarding Registration to Asterisk
by Wilkins, Steve 07 Dec '17

07 Dec '17

Hello All, Problem: My main issue is that when I "REGISTER" a client via Kamailio, and I attempt to "Dial" a different endpoint within an Asterisk Dial Plan, Asterisk throws an error stating that the endpoint (the number I am dialing via "Dial") is not registered or reachable. However, commands like "Playback" do work correctly for the client I "REGISTERED" via Kamailio. E.g. I register client 10001 in Kamailio, I then register another client 10002 in Kamalio; both 10001 and 10002 can exercise an Asterisk Dial Plan which will play videos/audio (No Problem). But, now I want 10001 to Call (Dial) 10002; it is at this point that Asterisk throws the error "10002 is not registered or reachable". I have tried many of the suggestion on many different help boards (several times) but I am still unable to forward a registration from Kamailio to Asterisk. With my current Kamailio configuration (I do use dispatching), I see , via tcpdump, Asterisk receiving a "REGISTER" request, and Asterisk sends back the "unauthorized" as expected, however, Kamailio does not re-send the "REGISTER" as is customary. I am not sure of the next step to take, but I feel I have a couple of options. * I can continue to try and figure out why Kamailio is not sending the second "REGISTER" (I have not yet been able to figure this out). * Tell Asterisk to not require authentication. (I am using pjsip and do not know how to not require authentication in Asterisk when the request is from Kamailio). I have put a lot of time into this one, and I am at a sticking point. Any help or suggestions would be very much appreciated. Thank you,

5 12

400 Bad From URI reply from kamalio
by Konstantin Tumalevich 07 Dec '17

07 Dec '17

One of our clients expiriences strange error. Their system sends BYE with different from address (which IP is presented in Contact header of initial INVITE) and Kamailio immediately responded with "400 Bad From URI". In which module can be generated this response? We use 4.3 version. -- Skype: konstantin.tumalevich

2 2

Question about parsing a jsonrpc_exec result with a colon in object name with jansson_get
by jenus＠cyberchaos.nl 07 Dec '17

07 Dec '17

Hi all, I'm trying to parse the result of a json rpc call to get the register statistics. # Get registrar stats jsonrpc_exec('{"jsonrpc": "2.0", "method": "stats.get_statistics", "params": ["accepted_regs"], "id": 1}'); jansson_get("result[0]", $jsonrpl(body), "$var(registrar_accepted)"); xlog("LOG_LOCAL3","L_WARN","$jsonrpl(body)\n"); The $jsonrpl(body) var contains: {"jsonrpc":"2.0","result":["registrar:accepted_regs = 2"],"id":1} With this code the result stored in $var(registrar_accepted) is : "AcceptedRegistrations":"registrar:accepted_regs = 1" I'm trying to get only the value. I have tried: jansson_get("result[0].registrar:accepted_regs", $jsonrpl(body), "$var(registrar_accepted)"); jansson_get("result[0].['registrar:accepted_regs']", $jsonrpl(body), "$var(registrar_accepted)"); jansson_get("result[0]['registrar:accepted_regs']", $jsonrpl(body), "$var(registrar_accepted)"); But this does not work. Do i need to escape the colon in the object name "registrar:accepted_regs" ? Thanks, Jan

1 1

Doing a quick test with http_async_client module from master branch
by Daniel-Constantin Mierla 07 Dec '17

07 Dec '17

Hello, anyone around here that can do a quick test with http_async_module? I don't have a testbed for it right now and I would spend the time on other resources if someone here can run some tests with http_async_client module from master branch. I did a bit of internal refactoring in order to be able to export the function to kemi interface -- nothing radical, but it would be good to know it didn't break anything. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

2 3

IMS with online Charging system
by Amar Tinawi 06 Dec '17

06 Dec '17

Hello all Is there any open souce ocs can be integrated with kamailio IMS ? Is it possible to integrate with CGrates ?

2 1

t_set_fr behaviour
by Kelvin Chua 06 Dec '17

06 Dec '17

hi guys, has anyone of you tried playing around with a scenario similar to this? A. invite -> t_set_fr() 2 seconds - if "100 trying" not received in 2 seconds, timeout. works fine. B. after receiving "100 trying" received, t_set_fr() 30 seconds - if 18x not received in 30 seconds, timeout. works fine C. after receiving the first 180, t_set_fr() 10 seconds - if "200 ok" not received in 10 seconds, it will not timeout, but instead it will timeout 30 seconds after B. i noticed this behavior recently and noticed that when a previous t_set_fr() is bigger than the new one, it will be ignored. so if i did 2 -> 10 -> 30, it works (swapping timeout values of B and C) my question is, is this an expected behavior? Kelvin Chua

2 2

load testing ims_charging
by Kelvin Chua 06 Dec '17

06 Dec '17

i am trying to load test the ims_charging module and notice some issues whenever i send test calls one by one, it's perfect. but when i use sipp and generate a bunch of calls, i would get this intermittently CRITICAL: ims_charging [ro_timer.c:122]: insert_ro_timer(): Trying to insert a bogus ro tl=0x7f207e91c2d0 tl->next=0x7f207e8b57b8 tl->prev=0x7f207e97c398 if i continue to hammer the server, it would end up failing all calls. When this issue happens, Ro_CCR will not return any value and will also not trigger the async route, looks like it just sits there frozen. during this time, sip client will receive a 408. also notice cdp complaining of latency usually around 1 second, using packet capture i am sure diameter server is responding properly. i think there is a choke point somewhere. i tried playing around with async_workers, no improvement. any pointers? Kelvin Chua

1 0

Re: [SR-Users] Kamailio Multi IP
by Duarte Rocha 06 Dec '17

06 Dec '17

Thnaks for the reply. I don't need mhomed since i do the socket forcing by myself after the the DISPATCH selects the Gateway. What i need is something that modifies the SIP Request in order to use the outbound IP in the R-URI, To-Header, etc. I'm doing that using the $td and $rd pseudo-variables, but i'm not sure if that is the safest approach. The TOPOS module is really usefull for this, thanks for mentioning. However, the dialog_expires parameter is really confusing, since it imposes a maximum time allowed for a dialog to exist (After the records are deleted, routing no longer works). I could work with a very long timer for this if the ended dialogs would get removed from the table automatically, but that doesn't happen. Cheers

1 0

redis via haproxy
by Aidar Kamalov 06 Dec '17

06 Dec '17

Hello, Kamailio after restart(or start) sometimes can't connect to redis (via haproxy) with logs: Dec 6 07:29:30 siptest /usr/sbin/kamailio[47291]: NOTICE: <script>: ROOT: KDMQ usrloc [764c076919c73540-39925(a)192.168.150.225] Dec 6 07:29:31 siptest /usr/sbin/kamailio[47263]: NOTICE: <script>: ROOT: OPTIONS <null> [0b4dc11225a6fc77294de97578061844@192.168.150.231:5060] Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: ndb_redis [redis_client.c:130]: redisc_init(): error communicating with redis server [redis] (127.0.0.1:6380/0): Resource temporarily unavailable Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: ndb_redis [ndb_redis_mod.c:122]: child_init(): failed to initialize redis connections Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: <core> [core/sr_module.c:923]: init_mod_child(): Error while initializing module ndb_redis (/usr/lib64/kamailio/modules/ndb_redis.so) Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: ERROR: <core> [core/pt.c:334]: fork_process(): init_child failed for process 5, pid 47269, "udp receiver child=4 sock=192.168.8.213:5060" Dec 6 07:29:31 siptest /usr/sbin/kamailio[47269]: CRITICAL: <core> [main.c:1611]: main_loop(): Cannot fork Dec 6 07:29:31 siptest /usr/sbin/kamailio[47252]: ALERT: <core> [main.c:740]: handle_sigs(): child process 47269 exited normally, status=255 Dec 6 07:29:31 siptest /usr/sbin/kamailio[47362]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 13 Redis and haproxy working fine (I test it by many attepts - echo ping|ncat localhost 6380) At tcpdump all attemps from kamailio have: *1 $4 PING +PONG *2 $6 SELECT $1 0 +OK my kamailio 5.0.4 config: modparam("ndb_redis", "server", "name=redis;addr=127.0.0.1;port=6380") my haproxy 1.5 config: backend bk_redis mode tcp option tcp-check tcp-check connect tcp-check send PING\r\n tcp-check expect string +PONG tcp-check send QUIT\r\n tcp-check expect string +OK server sipkazan 192.168.107.195:6379 check inter 1s server local 127.0.0.1:6379 check inter 1s backup -- Aydar A. Kamalov

2 2

Re: [SR-Users] using bcrypt passwd hashing
by Walter Martín Villalba 06 Dec '17

06 Dec '17

Hello everyone, and thanks very much for your feedback. Some responses and further questions below. Daniel> Latest kamailio versions support also SHA256 algorithm Martín> SHA256 is also a bad choice for storing passwords. See details here: https://crackstation.net/hashing-security.htm Daniel> However, the main blocker in suing a different hashing algorithm are the sip client devices (mainly hardphones), which implement only MD5. If you implement your own client app, then you can extend kamailio to support whatever hashing you do in the client. Then, of course you can use client side tls certificates for authentication, which should be better than any hashing algorithm. Martín> I do implement my own client app, even though I use a third party SIP stack, which currently doesn't support any other auth methods besides basic and MD5 (standard ones). I am planning to send username and passwd as custom SIP headers in the REGISTER message, probably encrypted, and this will travel on top of TLS. Then Kamailio can extract these custom headers and call a custom python script to decrypt the values and do the authentication (bcrypt password and compare with the one in database). Client certificates are good but only in certain situations (e.g. not if you want a zero footprint client such as a web-based client), and in most cases a pain to manage when your user base grows. Alex> Do you know of any mainstream SIP UACs which support anything other than standard MD5 digest auth? Martín> I don't, but haven't really worked much at all with 3rd party SIP clients. I doubt there's any support for newer passwd hashing schemes, unfortunately. ---------- Now the details.... I'm looking at sipcomm.cfg and see it calls www_authenticate (defined in modules/auth_db/authorize.c). I believe I would need to create a similar function, e.g. bcrypt_authenticate, and call this instead, with the username and passwd values I get in my custom headers (as explained above). The routine would decrypt the values, look up the user in the database, bcrypt the passwd extracted from the custom header, and compare with the one in the database. Doesn't sound too hard, but I do have some concerns related to other functions that www_authenticate may be doing, that I would also need to do in my bcrypt_authenticate function in order to keep Kamailio functioning properly. For example, www_authenticate could be changing some values in the database and/or other temporary storage. I took a quick look at the implementation and tried to follow the calls inside it. I see calls to mark_authorized_cred, check_auth_hr (or auth_check_hdr_md5), and generate_avps, and that some of these functions are indeed changing some values here and there. So, before spending more time looking into these details, I wanted to see if any of you have any suggestions about how to handle this situation, i.e. maybe all I need to do in bcrypt_authenticate is to check the credentials and then set one flag in the database for the user that was just authenticated? Does the explanation above make sense to you? Please let me know any suggestions or further guidance you may have. Thanks a lot, Martín. On Mon, Nov 13, 2017 at 3:00 AM, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Branch 5.1 created (Daniel-Constantin Mierla) > 2. Development open in master branch (to be v5.2.x) > (Daniel-Constantin Mierla) > 3. Re: using bcrypt passwd hashing (Daniel-Constantin Mierla) > 4. Re: t_set_fr behaviour (Daniel-Constantin Mierla) > 5. Re: t_set_fr behaviour (Daniel-Constantin Mierla) > 6. Re: AVPOPS: is_avp_set/avp_check "name" parameter as > variable. (Daniel-Constantin Mierla) > 7. Re: strange --dialog in delete state is too old-- log line > managing dialog hashes (Daniel-Constantin Mierla) > 8. Re: 183 acc records even if early_media equals to 0 > (Marco Capetta) > 9. Re: Kamailio issue (Daniel-Constantin Mierla) > 10. Re: using bcrypt passwd hashing (Yuriy Gorlichenko) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 12 Nov 2017 14:42:35 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org>, > "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org > > > Subject: [SR-Users] Branch 5.1 created > Message-ID: <dca81faa-dec2-4e12-704f-b382d23493d7(a)gmail.com> > Content-Type: text/plain; charset=utf-8 > > Hello, > > the GIT branch 5.1 has just been created, it will host the release > series 5.1.x. To get this branch from GIT, you can use: > > > git clone https://github.com/kamailio/kamailio.git kamailio > cd kamailio > git checkout -b 5.1 origin/5.1 > > > Hopefully in two-three weeks time frame the full release of 5.1.0 will > be out. > > >From now on, any corresponding fix has to be pushed first to master > branch and then cherry-picked to branch 5.1. No new features can get in > branch 5.1. Enhancements to documentation or helping tools, as well as > kemi exports are still allowed. If you are not sure about doing or not a > backport, ask on sr-dev mailing list. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > > > ------------------------------ > > Message: 2 > Date: Sun, 12 Nov 2017 14:50:45 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Devel Mailing List" <sr-dev(a)lists.kamailio.org>, > "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org > > > Subject: [SR-Users] Development open in master branch (to be v5.2.x) > Message-ID: <07baf03f-0d1b-30f6-45d2-cacfc3dfec99(a)gmail.com> > Content-Type: text/plain; charset=utf-8 > > Hello, > > git branch 5.1 was just created (to host the release series v5.1.x), > therefore new features can now be pushed again in master branch. They > will be part of the next future release, likely to be numbered 5.2.x. > > Any fixes that affect existing code in branches 5.1 or older version > have to be backported - push first to master and then cherry pick -- see > the contributing guidelines at: > > - > https://www.kamailio.org/wiki/devel/git-commit-guidelines# > backporting_commits > > Many thanks to all contributors so far! Testing of branch 5.1 and giving > feedback for it is very appreciated! > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > > > > ------------------------------ > > Message: 3 > Date: Mon, 13 Nov 2017 09:22:17 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, Yuriy Gorlichenko < > ovoshlook(a)gmail.com> > Subject: Re: [SR-Users] using bcrypt passwd hashing > Message-ID: <c7bb57e5-16dd-f5c2-f4ac-e3060f3b45bb(a)gmail.com> > Content-Type: text/plain; charset="utf-8" > > > > On 12.11.17 10:33, Yuriy Gorlichenko wrote: > > You can realize any of auth methods by yourself and include it via > > config file/kemi on lua/by adding module > > > > forexample I added SSO auth without any troubles instead of basid MD5 > > for some projects. > Out of curiosity, what do you refer by SSO? > > Cheers, > Daniel > > > > 2017-11-11 18:49 GMT+03:00 Alex Balashov <abalashov(a)evaristesys.com > > <mailto:abalashov@evaristesys.com>>: > > > > Do you know of any mainstream SIP UACs which support anything > > other than standard MD5 digest auth? > > > > On November 10, 2017 7:11:26 PM EST, "Walter Martín Villalba" > > <wvillalba(a)gmail.com <mailto:wvillalba@gmail.com>> wrote: > > >Hello, > > > > > >I did some searches online and talked to some colleagues and it > seems > > >Kamailio only supports the traditional HTTP digest authentication, > > >which > > >uses MD5. I would like to know if any of you has been successful in > > >using > > >bcrypt/scrypt/pbkdf2 passwd hashing, instead of MD5, which has been > > >deemed > > >as obsolete and insecure a long time ago. Perhaps you've written > your > > >own > > >auth module, or just modified the config script to call some other > > >credential checking routine using a custom python/perl script (I'm > > >thinking > > >of doing the latter, of nothing better is available). > > > > > >If any of you have done something like this, using bcrypt or any > > other > > >current and secure hashing algorithm, I would appreciate some > > guidance. > > > If > > >you haven't, aren't you concerned about storing MD5 password > > hashes in > > >your > > >database? > > > > > >Note: if I can't find a good answer using this list, I will try the > > >developer's list next. > > > > > >Thanks in advance, > > > > > >Martín. > > > > > > -- Alex > > > > -- > > Sent via mobile, please forgive typos and brevity. > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> > > > > > > > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > >

1 0

Kamailio Multi IP
by Duarte Rocha 05 Dec '17

05 Dec '17

Greetings, I have a Kamailio that is hosted on multiple IP's. Each client connects to an IP and there is another IP that communicates with the PSTN. The problem is that I don't want any client "seeing" any IP besides his. If the client (connected to kamailio's IP 1.1.1.1) receives an request coming from the PSTN (connected to Kamailio's IP 2.2.2.2) , it will have 2.2.2.2 on the R-URI, To Header,etc. Is there a method to ensure kamailio will manipulate the packet and use the correct IP ? Currently i'm directly changing the R-URI and To Header, but I don't if it's the correct approach to this. I would also like to know the best way to do those changes right before the packet leaves Kamailio. I know that OnSend route doesn't allow changes in the buffer. I tried Branch routes but not every request goes through that. Thanks for the help. Cheers.

2 1

how to escape asterisk in pattern matching
by voipspace voipspace 05 Dec '17

05 Dec '17

how to escape asterisk in pattern matching if($rU=~"\d+*") regards

2 1

How to change To header URI from Kamailio?
by Arun NV 05 Dec '17

05 Dec '17

Hi, I am using *Kamailio 4.4*. I would like to forward the request to a different port number of my endpoint. I have changed the destination *URI *and the *INVITE *correctly reached to the new port. But the *To header *in the *INVITE *request has the old port. so the endpoint is not responding to the request. Then I tried to remove and replace *To header* using *remove_hf("To")* and* insert_hf("To: $var(modified_to_header) \r\n");* functions. But the to header not changed. So, is there any way to change the *To header URI*.

2 1

Dialog Module Hash Size vs Number of Dialogs
by Mark Blackford 04 Dec '17

04 Dec '17

Hello, I am trying to properly size the use of the Dialog Module hash for our implementation using: modparam("dialog", "hash_size", <number that is power of two>) However, in my testing, I have been unable to figure out the relationship between the hash size and a number of dialogs I need to support. I think the hash size is specifying a memory block in kB, but even setting the hash size to a very tiny number does stop me from creating hundreds of dialogs. Is there a way to determine a relationship between the hash size and a rough number of dialogs that would be expected? An example of a a dialog looks like this from kamctl: [root@kamailio01 ~]# kamctl dialog show dialog memory records dialog:: hash=22:70 state:: 4 ref_count:: 2 timestart:: 1512151205 timeout:: 36083666 callid:: 0gQAAC8WAAACBAAALxYAAClws2wyL8GE+CSgRY7HIhmg9ZUIISZad46ntOPng3iPIcLaxzLFaytRTI7M0Bzz0g--(a)10.155.8.40 from_uri:: sip:b53667d44239457fbc94fc2f4c4e25a6@sip.dcs-staging.net from_tag:: 10.155.8.40+1+689d7e5e+8fcf481a caller_contact:: sip:43f0ae1480846185e8803f21e9f2b721@10.155.8.40:5060 ;transport=udp caller_cseq:: 24115 caller_route_set:: caller_bind_addr:: udp:10.155.8.11:5060 callee_bind_addr:: udp:10.155.8.11:5060 to_uri:: sip:2052773090@sip.dcs-staging.net to_tag:: sip+1+bdcd0004+2038f37c callee_contact:: sip:ca2013e84f10348a1cc825c12562bde7@10.155.8.40:5060 ;transport=udp callee_cseq:: 0 callee_route_set:: Thanks! -- Mark Blackford Digium Cloud Services 678.230.8769

3 5

How to use ruri-pattern with allow_trusted?
by voipspace voipspace 04 Dec '17

04 Dec '17

Hi Have used allow_trused to pass the allowed ip. Now I want to check the request has a correct tech prefix. With an entry in ruri-patten column in the trusted table for example 75622#. So if calls are to olny be allowed when the "to" destination starts with 75622#. Regards

1 0

max_contacts does not work as expected
by Gholamreza Sabery 04 Dec '17

04 Dec '17

I have a Kamailio server that has: modparam("registrar", "max_contacts", 2) But recently I noticed some users have more than two records in the location table. For example one of the users has three different records with nearly the same information. The records are as follows. Is this behavior normal? (I am using Kamailio 4XX). *************************** 1. row *************************** id: 171296745 ruid: uloc-2c69-59db6c73-7ca-132d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: XpCnmaAhRF cseq: 43 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 14 *************************** 2. row *************************** id: 171296746 ruid: uloc-2c69-59db6c73-7f2-eb5d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: BzMKuyJN8C cseq: 50 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 23 *************************** 3. row *************************** id: 171296747 ruid: uloc-2c69-59db6c73-80f-541d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: 3p9P4PgZSv cseq: 51 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 15

4 12

Dialog Module Hash Size vs Number of Dialogs
by Mark Blackford 04 Dec '17

04 Dec '17

Hello, I am trying to properly size the use of the Dialog Module hash table for Kamailio 4.1.3 implementation using: modparam("dialog", "hash_size", <number that is power of two>) However, in my testing, I have been unable to figure out the relationship between the hash size and a number of dialogs I need to support. I think the hash size is specifying a memory block in kB, but even setting the hash size to a very tiny number does stop me from creating hundreds of dialogs. Is there a way to determine a relationship between the hash size and a rough number of dialogs that would be expected? Thanks, -- Mark Blackford Digium Cloud Services 678.230.8769

2 1

How to get call on an extension, which is registered when a call towards it reach kamailio. How to modify pending session parameters in kamailio?
by Arun NV 04 Dec '17

04 Dec '17

Hi, I am using *Kamailio 4.4* as the proxy with my *Asteri**sk *server. I am trying to develop a scenario where an extension gets registered on asterisk via *Kamailio *when it receives a push notification. This push notification is sent to the sip extension when a call towards this extension reaches to the *Kamailio*. For example, suppose there is two SIP extension(* extension 1 and extension 2*) registered on *Asterisk *via Kamailio. When a call from *extension 1* reaches the asterisk, it forwards the *INVITE *request towards *extension 2* via Kamailio.Kamailio will try to forward it to extension 2. suppose the *extension 2* is not able to receive the *INVITE *request from *Kamailio*. When extension two receive a push notification, it will register on asterisk. So I need to get the call on *extension 2 *through the new registration. We are trying to simulate registration of extension to the asterisk when receiving the push notification. First, we registered *extension 2 *and disconnected the network. Then we tried to register the same extension when a call from *extension 1 *reaches to Kamailio. This is a simulation of push-based registration since an extension that receives the push will attempt to register when an incoming call is received. When asterisk sends *INVITE *request to Kamailio, it immediately responded with *100 trying* provisional response. This *100* response by Kamailio towards asterisk prevents asterisk from re-transmitting the INVITE. Then Kamailio tried to send and retransmit the packet to *extension 2*, which does not have network access. This* extension 2 *was on port number *24071*. Even after successful registration(in port *59995*) of the *extension 2*, Kamailio continued to transmit the packets to the old port. After that, we have configured Kamailio in a way that it won't send an immeadeate provisional reply(*100 trying* ) for *INVITE *request. Here Kamailio is not immediately sending *100 trying* message to *Asterisk*. This forces Asterisk to re-transmit. Asterisk was found to retransmit the same packets. However, even after the successful registration of* extension 2*, asterisk continued to send the old invite to Kamailio not the new one to the latest port. This is the problem for me since push relies on the *INVITE *reaching the phone at the correct port number. So, is there other good approaches to solve this issue? One thing I would like to try is modifying the pending *INVITE *request towards old registered port with the new port details when new registration reaches to Kamailio. Can I get the ongoing requests from Kamailio? Please suggest a viable solution. Thank you, Arunbalan

2 1

Using dispatcher's algorithm 12
by George Diamantopoulos 04 Dec '17

04 Dec '17

Hello all, I've been trying to use dispatcher's algorithm "12" to fork REGISTER messages constructed with uac module to all destinations within a set. However, the request is routed to only one destination. Is anything else needed in addition to calling ds_select_dst("0", "12") in the configuration script? Thanks. BR, George

3 5

bash script to install siremis&kamailio
by Atux Atux 04 Dec '17

04 Dec '17

Hi. i would like to ask if someone has found any bash script that could install all at once for Kamailio and Siremis in ubuntu/debian or even in centos.

2 1

dmq issues
by Aidar Kamalov 03 Dec '17

03 Dec '17

Hello. I have 4 kamailio nodes with same config: modparam("dmq", "server_address", "sip:LOCALIP:5050") modparam("dmq", "notification_address", "sip:192.168.8.213:5050") modparam("dmq", "notification_address", "sip:192.168.10.54:5050") modparam("dmq", "notification_address", "sip:192.168.107.195:5050") modparam("dmq", "notification_address", "sip:192.168.150.225:5050") modparam("dmq", "num_workers", 6) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "batch_size", 2000) modparam("dmq_usrloc", "batch_usleep", 1000) regularly (don't know why) I've got issues - replication is broken, for example now at all nodes I have: # kamcmd dmq.list_nodes { host: 192.168.8.213 port: 5050 resolved_ip: 192.168.8.213 status: 2 last_notification: 0 local: 0 } { host: 192.168.10.54 port: 5050 resolved_ip: 192.168.10.54 status: 2 last_notification: 0 local: 0 } { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 8 last_notification: 0 local: 0 } { host: 192.168.107.195 port: 5050 resolved_ip: 192.168.107.195 status: 2 last_notification: 0 local: 1 } but at node 192.168.150.225: # kamcmd dmq.list_nodes { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 2 last_notification: 0 local: 0 } { host: 192.168.150.225 port: 5050 resolved_ip: 192.168.150.225 status: 2 last_notification: 0 local: 1 } nodes restart doesn't help. but if i exec systemctl stop kamailio && sleep 15 && systemctl start kamailio it is partially solve my issue - dmq.list_nodes show all nodes, but at 192.168.150.225 i see not all user locations: they are added gradually There is no errors in logs Dec 1 08:28:37 sipufa /usr/sbin/kamailio[18252]: NOTICE: <script>: ROOT: KDMQ notification_peer [15f3edf9258014d1-7114(a)192.168.8.213] Dec 1 08:28:38 sipufa /usr/sbin/kamailio[18271]: NOTICE: <script>: ROOT: KDMQ notification_peer [7acb9eae627482f8-87828(a)192.168.10.54] Dec 1 08:28:39 sipufa /usr/sbin/kamailio[18250]: NOTICE: <script>: ROOT: KDMQ notification_peer [621ad9f474b335e2-18272(a)192.168.150.225] Dec 1 08:28:40 sipufa /usr/sbin/kamailio[18252]: NOTICE: <script>: ROOT: KDMQ notification_peer [7b817f6a6244f923-8872(a)192.168.107.195] Dec 1 08:28:47 sipufa /usr/sbin/kamailio[18271]: NOTICE: <script>: ROOT: KDMQ usrloc [7acb9eae627482f2-87794(a)192.168.10.54] Dec 1 08:29:20 sipufa /usr/sbin/kamailio[18251]: NOTICE: <script>: ROOT: KDMQ usrloc [7acb9eae627482ef-87790(a)192.168.10.54] -- Aydar A. Kamalov

3 8

kamailio as application server
by eyas barhouk 01 Dec '17

01 Dec '17

Hello if i need to use kamailio as an application server like mmtel for example how could i do that and what is the features that kamailio have if i want to use it as application server in another word what is the type of application server it could be act like (mmtel - presence) ? and thanks for help

2 1

In need of a simple, stateless 302 redirect for INVITE
by Robert 01 Dec '17

01 Dec '17

Hello again, I am not a SIP expert, and I’m new to Kamailio, really enjoying learning it, but I’ve hit a blocker and I’m pretty sure I’ve missed something really simple. I’m looking to implement the Redirect Server from here: https://tools.ietf.org/html/rfc3665#section-3.6 <https://tools.ietf.org/html/rfc3665#section-3.6> (the Avaya Session Manager being Alice), Kamailio as the Redirect Server. My configuration is working fine, and I can see the 302 going back (to Avaya SM), I then get the ACK I’m expecting. The incoming SIP request contains a number of fields, including a Contact header, which I’m told SM needs on the return. But the send_reply() or sl_send_reply() doesn’t return the Contact header, and I am unable to manually set it, or any other header, using any of the append_hf(), append_to_reply() etc. mechanisms. I’m executing a msg_apply_changes() which succeeds, but no Contact header is returned. I’ve tried an sl_send_reply(“100”, “Trying”) before any other processing, and Wireshark confirms that the SIP response is basically identical to the 302 response sent after processing… The goal is to run this stateless, I’m assuming this is possible - it certainly feels like it should be. I’ve re-read the intros & tutorials, and other reading material such as this http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf <http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf> but, this, even my best Google-fu has left me utterly stumped… The sl_send_reply documentation https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply <https://kamailio.org/docs/modules/5.2.x/modules/sl.html#sl_send_reply> states “If the code is in the range 300-399 (redirect reply), the current destination set is appended to the reply as Contact headers” but the Contact header is definitely not being set when I use this. Apologies in advance if eyes are rolling and this is really obvious, or (and?) that my lack of SIP knowledge is shining through - any tips or guidance would be appreciated. I certainly feel overwhelmed by the number of modules and the interplay between them. If the answers to this, and other things along these lines are contained in the admin book, that’s fab, I’ve just filled the form out :) Cheers - Robert...

3 3

DBURL password in clear
by Robert 01 Dec '17

01 Dec '17

Hello, I’m working for a UK high street bank and our Kamailio implementation has been challenged because we’ve got database passwords held in clear in the configuration file. I am unable to find any examples of where this has been worked around, there doesn’t seem to be any module or configuration means of supplying a variable in the modparam() entry that is expanded a startup. The security tutorials only seem to relate to the SIP level of security, not Kamailio as a platform. My requirement is simple, I need to be able to supply a password via means such as loading a variable from a run-once script at start up, or a module. The ideal would be to be able to read in a Docker secret :) I am by no means a Kamailio expert, so apologies in advance if this is a mindblowingly basic thing to achieve, but I do feel I’ve exhausted the Kamailio documentation, wiki etc. and all the goodness Google usually has to offer and drawn a blank. Sincere thanks in advance for any assistance. Cheers - Robert...

5 14

Dispatcher latency stats monitoring with statsd
by Julien Chavanton 01 Dec '17

01 Dec '17

Not sure if anyone is using "latency measurement" feature in Kamailio dispatcher module, I thought I could share a very simple Python script that I am using to monitor remote gateway using this. Sample Script is attached, it will post data to statsd and is working in cron The feature can be enabled : 3.27. ds_ping_latency_stats (int) Enable latency measurement when pinging nodes If set to 0, disable latency measurement. If set to 1, enable latency measurement https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc… This is one use case of this feature, another one will be to load balance based on congestion detected on latency, I am planning to share a lab experiment in a little while. Regards

2 1

[CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 30 Nov '17

30 Nov '17

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2018 takes place 3-4 February 2018 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 4th of February 2018. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 30th of November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM18/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2018/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 3rd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on the 2nd of February 2018 http://wiki.xmpp.org/web/Summit_22 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 2nd of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

3 2

blf in kamailio dmq cluster
by Aidar Kamalov 30 Nov '17

30 Nov '17

Hello, I have two(or more) kamailio servers with dmq module and dmq_usrloc - sip1.example.com and sip2.example.com publish and subcribe replication works too. BLF works fine if clients connected to the same kamailio server, because in active_watchers table stored with same domain sip1.example.com But what to do when call going from one server to second? Is there any configuration to ignore domain in pusblish, subscribe, notify, etc? -- Aydar A. Kamalov

3 7

OpenSIPS-devel not starting (TCP Error)
by SamyGo 29 Nov '17

29 Nov '17

Hi, I just installed the head version from git. version: opensips 2.4.0-dev (x86_64/linux) flags: STATS: On, SHM_EXTRA_STATS, EXTRA_DEBUG, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. git revision: d1a4419 main.c compiled on 19:53:20 Nov 28 2017 with gcc 4.8 It was compiled with EXTRA_DEBUG and DBG_TCPCON flags; While starting opensips with the residential vanilla script it gives following error: Nov 28 20:05:52 [30899] ERROR:core:tcp_init: oom con hist Nov 28 20:05:52 [30899] CRITICAL:core:main: could not initialize tcp Nov 28 20:05:52 [30899] INFO:core:cleanup: cleanup Nov 28 20:05:52 [30899] DBG:core:shm_mem_destroy: destroying the shared memory lock Nov 28 20:05:52 [30899] NOTICE:core:main: Exiting.... already running Even if I remove the proto_tcp, and tcp related things this still gives error and doesn't start. Kindly help. Regards, Sammy

2 3

parse_from_header(): bad From header
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. We experience the problem with malformed messages. First of all, I want to say, that most of time kamailio works well and nothing happens that can drop sessions. But from time-to-time something changes (may be in the network) and kamailio receives requests with malformed headers (To or From hfs). The schema: uplinks -> kamailio <-> routing server (asterisk) Malformed messages were received only from asterisk server. Sip debug on asterisk showed that messages were transmitted correctly and headers were nice. But kamailio obtains changed (broken) packets or perhaps can't read them properly. How does it look like: Nov 10 12:37:06 kamailio-name kamailio[965]: INFO: <script>: Going to NATMANAGE for BYE from sip:useragent7122@kamailio_address:5068 (IP=wss:client_address:62881) - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: NOTICE: <script>: Relaying request to <null> - R=sip:dialed_did_service@asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[965]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: dialog [dlg_cseq.c:89]: dlg_cseq_prepare_msg(): cannot parse FROM header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: uac [replace.c:783]: restore_uris_reply(): failed to find/parse FROM hdr Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: Skip manage DEVICE_STATE for BYE from <null> (IP=udp:asterisk_address:50600) with (200 - OK) - R=<null> ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: INFO: <script>: ------------------------------------MANAGE BYE by DEVICE_STATE_BYE_MANAGE route - 200 - OK M=BYE IP=udp:asterisk_address:50600 ID=1b0044070de406153bf0e4b84d6bb793@asterisk_address:50600 Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine_funcs.c:331]: get_from_tag(): failed to parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: rtpengine [rtpengine.c:2252]: rtpp_function_call(): can't get From tag Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: [<sip:useragent6] . Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: <core> [parser/parse_from.c:75]: parse_from_header(): bad From header [<sip:useragent6<+.w>;tag=jja7l45gd1] Nov 10 12:37:06 kamailio-name kamailio[937]: ERROR: pv [pv_core.c:465]: pv_get_from_attr(): cannot parse From header How the packet looks like: ACK sip:tt7etmau@bb0pd5t63b26.invalid;alias=client_address~57573~6;transport=ws SIP/2.0 Via: SIP/2.0/UDP asterisk_address:50600;branch=z9hG4bK117106ca;rport Route: <sip:kam2.domain.com:5068 ;nat=yes;transport=udp;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331>,<sip:kam2.callision.com:5068 ;nat=yes;transport=ws;r2=on;ftag=as06869a5d;lr=on;vsf=AAAAAEZZSVVDVV1VRBlUUkUpFQxdVUZYXiRCHU1RQl1HRVNaXhhAQ3RuLnR3aWxpby5jb20-;vst=AAAAAAAAAAAAAAAAAAAAAAAfBwQRQE8NBR9BChIFAwddCgADFEVDQlYudHdpbGlvLmNvbQ--;did=b0c.7331> Max-Forwards: 70 From: "19172423539" <sip:19172423539@asterisk_address:50600>;tag=as06869a5d To: <sip:useragent7122@10.0.1.18:5068>;tag=nmv7kmpau3 Contact: <sip:19172423539@asterisk_address:50600> Call-ID: 629d5a132c194c536b5f5c1a2a3c6e32@asterisk_address:50600 CSeq: 102 ACK User-Agent: asterisk PBX Content-Length: 0 The transport is used to send messages between asterisk and kamailio is TCP. TCP configurations: tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 I found the idea, that we need to change following parameters: modparam("uac","restore_mode","auto") modparam("uac","restore_dlg",1) to: modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) I did that, now it looks like all is fine, but I think it can get back. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua>

3 5

Source socket/port for in-dialog requests
by Donat Zenichev 29 Nov '17

29 Nov '17

Hi community. My apologies for so frequent appealing to you. I'm trying to solve problem with ending of sessions. The problem consists of no 200 OK coming from uplink on our BYE requests. Topology. First leg: Webrtc client <-wss-> kamailio <-sip tcp-> asterisk routing server Second leg: Uplink <-sip tcp-> kamailio <-sip tcp-> asterisk routing server The problem appears only in case when dialog was ended by webrtc client. The first leg (dialog) of the call ends nice, without any hints on problem. But the second leg (with uplink) has problem with no 200 OK (coming from uplink) on BYE request coming from asterisk. Handshake between asterisk server and uplink establishes properly. It looks like: uplink.domain.com:5060 -> INVITE tcp -> our.kamailio.server:5060 -> INVITE tcp -> our.asterisk.server:5060 . uplink.domain.com:5060 <- 100 trying tcp <- our.kamailio.server:5060 . uplink.domain.com:5060 <- 180 ringing tcp <- our.kamailio.server:5060 <- 180 ringing tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 <- 200 OK tcp <- our.kamailio.server:5060 <- 200 OK tcp <- our.asterisk.server:5060 . uplink.domain.com:5060 -> ACK tcp -> our.kamailio.server:5060 -> ACK tcp -> our.asterisk.server:5060 . uplink.domain.com <-media stream-> rtpengine <-media stream-> our.asterisk.server . Here kamailio starts to use random source port for relaying in-dialog BYE uplink.domain.com:5060 <- BYE tcp <- our.kamailio.server:45355 <- BYE tcp <- our.asterisk.server:5060 . And here the leg with uplink is expected to end with 200 OK (coming from uplink proxy). But uplink doesn't answer at all. We requested a tcpdump from uplink to see how packets are forwared from their side. And I saw that that 200 OK tries to be sent within first tcp session - to 5060 port of kamailio server. But sngrep on our side shows nothing, nothing appears in kamailio log, so 200 OK can't reach the 5060 socket because of some transport problem I think. Top via hf of our BYE request, contains kamailio record = uplink.domain.com:5060 My main thought on this count is to suppress using of random source ports for in-dialog requests, this behaviour looks irrelevant. We have turned on mhomed parameter (mhomed=1). And cookbook says: "Set the server to try to locate outbound interface on multihomed host. This parameter affects the selection of the outgoing socket for forwarding requests." But, there is a big but - we can not turn of this parameter for this moment, because routing script made with using of this one. Kamilio works on AWS EC2 platform. It has following IP address schema: private address atouched to container -> public address assinged to this private address (so this is standard ip address schema for AWS containers). For this example private address will be: 10.0.0.1 and public address will be: 100.0.0.1 Configuration - main parameters (all values are changed for an example): advertised_address=our.kamailio.server advertised_port=5060 alias=our.kamailio.server alias=our.kamailio.server:5060 alias=100.0.0.1 alias=100.0.0.1:5060 alias=10.0.0.1 alias=10.0.0.1:5060 auto_aliases=no port=5060 listen=100.0.0.1 listen=10.0.0.1 listen=tcp:100.0.0.1:5060 listen=tcp:10.0.0.1:5060 mhomed=1 fork=yes fork_delay=5000 children=6 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_connect_timeout=5 tcp_send_timeout=5 tcp_rd_buf_size=16384 tcp_keepalive=yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=15 tcp_max_connections=4096 One of the solutions to this problem was to add following inside the relaying route block: $fs = "tcp:" + "10.0.0.1" + ":5060"; if (!t_relay()) { sl_reply_error(); } But it seems to me it's not a smart solution. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 https://w-net.us/ <http://wnet.ua> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 1

Best way to do routing against a list of gateways
by voipspace voipspace 29 Nov '17

29 Nov '17

3 6

allow_trusted not passing
by voipspace voipspace 29 Nov '17

29 Nov '17

4 5

URL problems connecting to external (Azure) MySQL database
by Floimair Florian 29 Nov '17

29 Nov '17

Hi! I'm having troubles connecting Kamailio to an external MySQL database (on Microsoft Azure Cloud). Initially creating the database with kamctl worked fine, but accessing it at runtime via db_url fails. A somewhat unusual thing about the connection to a MySQL database on Azure is that the username contains an '@' character. This leads me to believe that this case is somewhat unhandled when parsing the database URI. Trying to connect to something like modparam("sipcapture", "db_url", "mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db") Results in errors starting Kamailio ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Unknown MySQL server host 'clouddb' (-2) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool ERROR: sipcapture [sipcapture.c:1146]: child_init(): unable to connect to database [mysql://user@clouddb:password@clouddb.mysql.database.azure.com/db] from capture_mode param. Is there a trick on how to achieve a connection in this case like some sort of escape character probably? Mit freundlichen Grüßen Florian Floimair (IMS) 1568

2 4

lcr_id vs lcr_count
by Patrick Wakano 29 Nov '17

29 Nov '17

Hello Kamailio list! I've been working with the LCR module and I wanted to use the lcr_id to match some ID's I have somewhere else. However, due to the lcr_count my lcr_ids must be less than the lcr_count. I understand the reason behind the lcr_count, what I think is a mistake is to have the lcr_id dependent of the lcr_count. Ideally we should be able to use any lcr_id value, just observing that the total amount of lcr groups should not exceed the lcr_count. The problem is this if that appears in the load_gws, to_gw and from_gw functions in lcr_mod.c: if ((lcr_id < 1) || (lcr_id > lcr_count_param)) { LM_ERR("invalid lcr_id parameter value %d\n", lcr_id); return -1; } I ended up using another table to make the conversion of ids, but this is horrible.... Any other reason to force the lcr_id to be less than the lcr_count? Best Regards, Patrick Wakano

3 3

how to add new media server in dispatcher table through "ANSIBLE"
by Narayan P 29 Nov '17

29 Nov '17

Hello Experts, I am new to ansible and deploying kamailio through ansible.Sucessfully deployed kamailio but my requirement is to add media server through ansible.As I am using dispatcher module in kamailio,not able to find any script that directly insert one media server IP and port into dispatcher table from ansible.(I am using ansible in one system and deploying kamailio in other). Any suggestion will be highly appreciated as always. Thanks, Narayan +91-9035438928

3 4

Re: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue.
by otron2016＠gmail.com 29 Nov '17

29 Nov '17

Through Firefox likely means via websockets (wss) rather than via the sip port. Many things could interfere with the latter. Take heart, though: websockets is harder to set up so you're likely through the worst. Look at logs and see what iftop shows you about traffic on the server as you try to make sip connections. Also try your sip client with a known-good sip server (e.g. Iptel) just to make sure the problem isn't with the client. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/28/2017 6:39 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Using SIP tools through Kamailio and Asterisk - chrome issue. Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won’t work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don’t know 100% for sure. Thank you

1 0

Using SIP tools through Kamailio and Asterisk - chrome issue.
by Wilkins, Steve 28 Nov '17

28 Nov '17

Has anyone seen the issue where, when using a sip tool and going from Kamailio to Asterisk, that audio and video won't work, yet it works fine through Firefox? I am almost sure this not an issue with Kamailio, but I am pretty new to SIP and don't know 100% for sure. Thank you

1 0

dmq replicate
by Aidar Kamalov 28 Nov '17

28 Nov '17

Hello, I set up kamailio cluster. dmq_usrloc works fine, but PUSBLISH and SUBSCRIBE replication not. If I make a call to first node I've got many(hundreds) PUBLISH messages from first node to first node and some messages (8 for example) to other nodes. And second xnotice in config not executes. And errors n log ERROR: <core> [core/sr_module.c:1726]: get_int_fparam(): Unsupported param type for int value: 32 How to replicate publish and subscribe, is it wrong way? my confiig: request_route { if(is_method("KDMQ")) { if ($Rp == 5050) { dmq_handle_message(); exit; } else { drop; } } ..... ..... if(is_method("PUBLISH|SUBSCRIBE")) { if(!dmq_is_from_node()) { xnotice("AAAAAAAA"); dmq_t_replicate("1"); xnotice("AAAAAAAABBBBBBBBBBB"); } } 2017/11/28 12:56:06.415601 192.168.8.213:5060 -> 192.168.8.213:5060 PUBLISH sip:34200@siptest. SIP/2.0 Via: SIP/2.0/UDP 192.168.8.213;branch=z9hG4bKc7a5.cee84051000000000000000000000000.0 To: <sip:34200@siptest> From: <sip:34200@siptest.>;tag=ae81df16b5a52fc1f62f9b45f337e660-db10 CSeq: 10 PUBLISH Call-ID: 4c8f9e6e799efcd7-57244(a)192.168.8.213 Content-Length: 556 User-Agent: kamailio (5.0.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Expires: 125 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:34200@siptest."> <dialog id="898932197-5060-86(a)BJC.BGI.D.DB" call-id="898932197-5060-86(a)BJC.BGI.D.DB" direction="initiator"> <state>Trying</state> <remote> <identity>sip:53003@siptest.t</identity> <target uri="sip:53003@siptest.tg"/> </remote> <local> <identity>sip:34200@siptest</identity> <target uri="sip:34200@siptest"/> </local> </dialog> </dialog-info> -- Aydar A. Kamalov

1 1

dispatcher/carrierroute and branches across SBC and remote peer
by Nathan Ward 28 Nov '17

28 Nov '17

Hi, We use Kamailio as a SIP server, and SEMS as an SBC. We have a need to use the dispatcher/carrierroute modules to provide HA/load sharing across two different parts of our environment: 1) Our SBCs 2) Our peer’s SIP servers We have pairs of SBCs (for now) and they have the same network connectivity - different IPs of course, but they can reach the same things. Our peer’s network has 2 SIP servers (again, for now) reachable through either SBC. We select which SBC pair to use based on where the calls are going, in a branch route. In some cases, we use parallel forking, in others we use serial forking (i.e. to a media server for a prompt, then sending the call to wherever it needs to go). Right now there is no automatic failover or load sharing, so this is all simple logic - if call is to x peer or y customer, use x SBC or y SBC. I’m trying to add the failover/loadbalancing now. I have a few half baked ideas about how to achieve a network like this, but thought I’d ask if anyone else had done this first before I went too overboard. Currently I’m thinking of calling cr_route to figure out which peer SIP server, from there figuring out which SBC pair, and then calling ds_select_dst to select which SBC in the pair to set $du. I’m being tripped up though, as I can’t call cr_route or ds_select_dst in a branch route, but, as far as I’m aware that’s where I’d need to do so in order to handle this for multiple branches. Should I be looping the request through the SIP server to achieve this (i.e. force $du = 127.0.0.1 in my branch_route), or, can I avoid that complexity? -- Nathan Ward

1 0

Kamailio didn't start before increasing fork_delay
by igor.potjevlesch＠gmail.com 27 Nov '17

27 Nov '17

Hello, We suddenly had an issue on one Kamailio instance: we were not able to restart. Kamailio started to boot and fork and suddenly crashed. The last logs reported a failure regarding the ability to connect to one MySQL instance. I finally succeed to restart after increasing: fork_delay=5000 to fork_delay=9000. How this could happen suddenly? We already restarted Kamailio on this server. Regards, Igor.

2 5

Does Kamailio support RFC4117?
by Ye, Charlie (NSB - CN/Qingdao) 27 Nov '17

27 Nov '17

Hi Friends, Does Kamailio support RFC4117? Do you have any suggestions if I want support RFC4117 through Kamailio? Thanks and Best Regards, Charlie

2 2

Dispatcher with kam & fs behind nat
by Daniel Greenwald 27 Nov '17

27 Nov '17

Dispatcher is sending inbound calls to Freeswitch boxes. Both the FS boxes and Kam are behind the same nat. What is the correct way to pass calls to the FS boxes so that all SIP between FS & Kam use local IP's? Should Kam put two record-route headers on, one with PUBLICIP and one with PRIVATEIP? How is this accomplished? Thanks

2 3

Kamailio 5.0 Android Calling Not working
by Zeeshan M 25 Nov '17

25 Nov '17

Hello Friends, I had successfully installed and configured Kamailio 5.0 on linux machine. And tried it with linphone app & following are the issues i am facing, 1) IOS to Android Calling not working with timeout error on server end when we checked logs 2) Android to Android calls also not working same error. Kindly let me know what could be the issue. Regards, Zeeshan.

1 0

Re: [SR-Users] sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

I just verified that I do have logging in all pertinent areas. I added tls and I do see "TLS accept successful" in the kamailio logs, however, I see no log being generated in route[xhttp:request], and I do have logging in the first line of this route. I don't think it is getting there. Also, I noticed I have event_route[xhttp:request] not route[xhttp:request]. Thank you, -Steve

1 0

sipML5 through kamailio
by Wilkins, Steve 24 Nov '17

24 Nov '17

Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen's in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve

2 5

Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

Just a guess but maybe later entries [like +HIGH:+MEDIUM:+LOW] put it back. Try switching the order so that !ECDHE and the others you're trying to exclude come after. Sent from Samsung Mobile -------- Original message -------- From: Ilyas Keskin <ilyask92(a)gmx.de> Date: 11/24/2017 10:19 AM (GMT-08:00) To: miconda(a)gmail.com,"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Cannot disable EC Diffie Hellman cipher suite Hi Daniel, yes I am using the tls.cfg file. I tried your suggestion to add the cipher suite string (notice the !EDCHE which I also added to the httpd ssl.conf) but nothing changed. [server:default] method = TLSv1 cipher_list = !DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem certificate = /etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem #ca_list = ./modules/tls/cacert.pem #crl = ./modules/tls/crl.pem Also here is a log snippet from tls module section of kamailio initialization. Notice first two lines. Also it seems to me the module actually ignores the local openssl installation and uses its own which has been compiled with the module itself (?). Other than that it seems to be accepting the cipher_list value just fine: Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:355]: mod_init(): With ECDH-Support! Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_mod.c:358]: mod_init(): With Diffie Hellman Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:587]: init_tls_h(): tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, compression: on Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_init.c:595]: init_tls_h(): tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, zlib compression: compiler: gcc -I. -I.. -I../include -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_ Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tls [tls_init.c:649]: init_tls_h(): tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 7864320 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 3932160 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_branch("MANAGE_BRANCH"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_reply("MANAGE_REPLY"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: WARNING: tm [tm.c:594]: fixup_routes(): WARNING: t_on_failure("MANAGE_FAILURE"): empty/non existing route Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSs<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSs<default>: certificate='/etc/letsencrypt/live/webrtc.ddnss.de/fullchain.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSs<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSs<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSs<default>: require_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSs<default>: cipher_list='!DH:!ECDHE:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSs<default>: private_key='/etc/letsencrypt/live/webrtc.ddnss.de/privkey.pem' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSs<default>: verify_certificate=0 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSs<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:670]: set_verification(): TLSs<default>: No client certificate required and no checks performed Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSc<default>: tls_method=12 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSc<default>: certificate='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSc<default>: ca_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSc<default>: crl='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSc<default>: require_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSc<default>: cipher_list='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSc<default>: private_key='(null)' Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSc<default>: verify_certificate=1 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSc<default>: verify_depth=9 Nov 24 18:56:20 kamailio-sip /usr/sbin/kamailio[2864]: INFO: tls [tls_domain.c:655]: set_verification(): TLSc<default>: Server MUST present valid certificate Would it be possible to compile the tls module with certain openssl config switches (i.e. no-ec no-dh)? Any other ideas? Best regards, Ilyas Keskin Am 24.11.2017 um 15:45 schrieb Daniel-Constantin Mierla: Hello, On 23.11.17 22:42, Ilyas Keskin wrote: Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. are you also using tls.cfg? If yes, there is an attribute for chiper list in it as well, try and see if works with it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Re: [SR-Users] sipML5 through kamailio
by otron2016＠gmail.com 24 Nov '17

24 Nov '17

"So, tls is required for wss?" Yes, as of the last rfc. It can work without it if the client software (browser) will accept it though by now most will treat it as insecure. I forget whether sipML5 will but my guess would be no. -------- Original message -------- From: "Wilkins, Steve" <swwilkins(a)mitre.org> Date: 11/24/2017 9:57 AM (GMT-08:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hello SamyGo, So, tls is required for wss? I thought that when I set a listen to Kamailio-Server-IPAddress:10443 and I sent a request through that IP and Port, that Kamailio would at least see the request and attempt to do something with it. I will go add tls and try again. Thank you! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Can you clarify the IP addressing scheme as you've mentioned. There is no TLS interface? Advertised address for 5060 but not for WSS interface. Both listen and advertise are public IPs ? Also,you're handling WSS requests. Do you've xlog ines in the route[xhttp:request] to view when a request lands.! My config has this: #!substdef "!MY_IP_ADDR!123.134.156.167!g" #!substdef "!MY_WS_PORT!6010!g" #!substdef "!MY_WSS_PORT!6011!g" listen=tcp:MY_IP_ADDR:MY_WS_PORT listen=tls:MY_IP_ADDR:MY_WSS_PORT Then the xhttp event route: ``` event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); #Deny any HTTP requests on any port other than WS/WSS ports. if ($Rp != MY_WS_PORT && $Rp != MY_WSS_PORT ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } #Handle HTTP(s) onwards. xlog("L_INFO", "HTTP Request Received\n"); ``` On Thu, Nov 23, 2017 at 12:18 PM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hi Sammy, First of all, thank you for taking the time to respond. Yes, port 10443 is opened. I have used this port before as asterisk’s WebRTC port and iptables shows it as open. No, I can’t even get a registration using the configuration I listed. I have an xdbg log statement right after the request_route, and I see nothing. I do know that my xdbg logs are working though because, if I register or make a call using any sip tool, I see all my logging and everything works correctly. -Steve From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of SamyGo Sent: Thursday, November 23, 2017 12:00 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] sipML5 through kamailio Hi Steve, Can you confirm that port 10443 is reachable behind the NAT to Kamailio server, validate iptables too Does your SIPml5 demo client register successfully to Kamailio? are there enough xlog lines to print out if anything lands in Kamailio. Regards, Sammy On Thu, Nov 23, 2017 at 11:34 AM, Wilkins, Steve <swwilkins(a)mitre.org> wrote: Hello, I am attempting to use sipML5 to test WebRTC. I have not been successful in getting messages through to Kamailio though. I am running Kamailio 5.0.3 on Cento 7. My listen’s in the kamailio configuration file are => listen=tcp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=udp:112.22.3.108:5060 advertise 34.226.187.61:5060 listen=tcp:112.22.3.108:10443 (which I will use in the sipML5 Expert mode) My sipML5 settings are => Public Identity - sip:user1@112.22.3.108 Realm - 112.22.3.108 Export mode setting are => WebSocket Server URL - wss://112.22.3.108:10443 (I have also tried wss://112.22.3.108:10443/ws) SIP outbound Proxy URL - udp://112.22.3.108:5060 (I have also left this blank) When I make a call I see no Kamailio activity (I have logging at the start of request_route) so I am not sure where the configuration error is. If I change the sipML5 configuration IP Address to use the asterisk IP Address, sipML5 works. My goal is to go WebRTC Client => Kamailio => Asterisk and eventually through some sort of media proxy. Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

Cannot disable EC Diffie Hellman cipher suite
by Ilyas Keskin 24 Nov '17

24 Nov '17

Hi there, I have set up a Kamailio 4.2.0 SIP server (centOS 7) for a university project regarding WebRTC comunication. While kamailio handles the signaling path I use the SIP.js demo phone js application (hosted on the same machine as kamaillio) for actual WebRTC stuff. For a deeper understanding and documetation purposes I have been trying to sniff the traffic with wireshark but failed due to the fact that kamailio uses Elliptic Curve Diffie Hellmann cipher suite (see wireshark snippet below) which is not decryptable. Secure Sockets Layer TLSv1.2 Record Layer: Handshake Protocol: Server Hello Content Type: Handshake (22) Version: TLS 1.2 (0x0303) Length: 89 Handshake Protocol: Server Hello Handshake Type: Server Hello (2) Length: 85 Version: TLS 1.2 (0x0303) Random: b8916e4e0f7c712503a77afcf4c9228598092c166353be50... Session ID Length: 32 Session ID: b0a31a6699a001b7991645dc61064ca4c4b073eff6913f26... Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) Compression Method: null (0) Extensions Length: 13 Extension: renegotiation_info (len=1) Extension: ec_point_formats (len=4) I already tried importing captured SSLKEYLOG pre master secret from chrome and private key file issued by letsencrypt without success. On top of that I set this line SSLCipherSuite !DH:!ECDH:!EDH:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL in /etc/httpd/conf.d/ssl.conf and compiled openssl with no-ec no-dh (which worked see below). [admin@kamailio-sip ~]$ openssl ciphers SRP-DSS-AES-256-CBC-SHA:SRP-RSA-AES-256-CBC-SHA:SRP-AES-256-CBC-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA:CAMELLIA256-SHA:PSK-AES256-CBC-SHA:SRP-DSS-AES-128-CBC-SHA:SRP-RSA-AES-128-CBC-SHA:SRP-AES-128-CBC-SHA:AES128-GCM-SHA256:AES128-SHA256:AES128-SHA:SEED-SHA:CAMELLIA128-SHA:IDEA-CBC-SHA:PSK-AES128-CBC-SHA:RC4-SHA:RC4-MD5:PSK-RC4-SHA:SRP-DSS-3DES-EDE-CBC-SHA:SRP-RSA-3DES-EDE-CBC-SHA:SRP-3DES-EDE-CBC-SHA:DES-CBC3-SHA:PSK-3DES-EDE-CBC-SHA [admin@kamailio-sip ~]$ Setting modparam("tls", "cipher_list", "AESCCM") (or different ciphers) in /etc/kamailio/kamailio.cfg seems to have no effect on the actual negoiated cipher suite. Am I missing something? Any help or pointers into the right direction will be much appreciated. Best regards, Ilyas Keskin

2 2

Kamailio KEMI Framework Tutorial
by Daniel-Constantin Mierla 24 Nov '17

24 Nov '17

Hello, I started a tutorial for Kamailio KEMI Framework written in markdown format targeting use of mkdocs. It is published on github as part of kamailio-docs repository: * https://github.com/kamailio/kamailio-docs/tree/master/kamailio-kemi-framewo… The mkdocs build (to browse and read) is online at: * http://kamailio.org/docs/tutorials/devel/kamailio-kemi-framework/ It is work in progress and contributions from the community are more than welcome -- doing push requests on github project is the recommended way for contributions: * https://github.com/kamailio/kamailio-docs/ The main goal is document the role of KEMI, its benefits and how it can be used. It addition it should cover the list of functions exported to KEMI by core as well as special KEMI functions. Regarding the functions exported by kamailio modules to KEMI I added as an skeleton the ones exported by acc module. However, I am not sure if we should do it there or add them to the existing docs of the modules. Opinions on this?!? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Request Time Out
by Vasim Antule 24 Nov '17

24 Nov '17

Hello, I am new to SIP/Kamailio , I am attempting to use Kamailio 5.0 in test environment. I am getting Request Time Out for Android Phones. Apple to Apple phones are working fine. But Android to Apple and Android to Android I am getting Request Time Out. Please Guide. Regards, Vasim Antule

1 0

Issue dumping an huge htable through xmlrpc
by Marco Capetta 24 Nov '17

24 Nov '17

Hello everyone, On Kamailio version 4.4.6 I have an issue dumping an huge htable through xmlrpc. We are using htables to store the subscriber registration attempts to prevent bruteforcing password attacks. Usually there aren't any issues to dump information using xmlrpc. On the contrary, if the number of records in the htable become huge (in the order of 10.000), xmlrpc output is truncated, therefore it is not possible to parse it. I have the same results either using a Perl library or the "xmlrpc_test.py" script included in package. I tried to increase the module parameters "binrpc_max_body_size" and "binrpc_struct_max_body_size" without success. Do you have any other suggestion? Thank you Regards Marco -- *Marco Capetta * Operations Engineer Sipwise GmbH <http://www.sipwise.com> , Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Phone: +43(0)1 301 2044 <tel:+4313012044> Email: mcapetta(a)sipwise.com <mailto:mcapetta@sipwise.com> Website: www.sipwise.com <http://www.sipwise.com> Particulars according Austrian Companies Code paragraph 14 "Sipwise GmbH" - Europaring F15 - 2345 Brunn am Gebirge FN:305595f, Commercial Court Vienna, ATU64002206

2 1

Failed to find command http_client_query
by Alexandru Covalschi 24 Nov '17

24 Nov '17

Hello everyone, Updated to 5.0.4 from repo, up-to-date debian 8, http_client.so is certainly loaded. When I try to change http_query to http_client_query it fails to load with error "failed to find command http_client_query". Any ideas? It is same on all my test servers. ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493

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msilo support for picture, audio and video?
by sscc 24 Nov '17

24 Nov '17

Hello Folks does msilo support picture, audio and video? regards sscc? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 5

Loadbalancing authenticated agetways
by SamyGo 23 Nov '17

23 Nov '17

Hi, I've a scenario where multiple mediaservers will be registered to Kamailio and for any incoming call from Upstream provider I've to perform load-balancing to the actively registered media-servers. dispatcher module does load-balancing for IP endpoints, is there any other module where I can perform load-balancing on registered clients(media-servers)? Thanks, Sammy

3 3

why do I need rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk via Kamailio, and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Thank you

3 2

rfc: clang format the code
by Daniel-Constantin Mierla 23 Nov '17

23 Nov '17

Hello, it was discussed during the last irc devel meeting and everyone there agreed to use clang-format to format the source code: - https://clang.llvm.org/docs/ClangFormat.html Is any developer (maybe not present during the irc devel meeting) opposing this? It is not going to be all at once, because some of the structures and initialization of them may be preferred to be kept with a custom format for being easier to read -- to have a section in the file not formatted, it has to be enclosed in between comments: /* clang-format off */ ... /* clang-format o. */ But for the rest of the code (i.e., functions) I think it will be better to agree on a format and try to use it. The clang-formatting should be done when one needs to update code in some files. Over the years, developers come and go, having each one with a custom formatting style is hard to maintain properly in long term. There are modules that do not indent coherently code inside blocks, using a mixture, more or less no style. This makes it hard for other developers to review in case of reported issues, but also for new comers to get familiar with the code. In the source tree there is already a .clang-format file which defines the options for formatting. It was used by me and other developers already -- I did it mainly in my modules or those components that do not have an active maintainer. But I plan to do it also on the components that have currently a maintainer at the moment I need to work on that code (e.g., fix some issue or export functions to kemi framework). My process of clang-formatting a module is like: - edit the files to add clang-format off/on comments around the parts I want to preserve as they are (typically mod exports -- params, functions, kemi -- and intialization of structures that are aligned to a specific indentation that makes it easier to read). - run: clang-format -i src/modules/modname/*.{c,h} - commit with these changes, no new feature or bug fix must be done in the same commit (can be combined with removing the history comments) If one wants different options inside .clang-format, please propose and let's discuss which one is preferred by the majority of the current developers. If no one shows anything against in a matter of few days, it will be considered accepted (after all, one can re-indent on own style if the rule is not accepted or overturned). Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Error when attempting to use rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

Hello, I entered this question before but did not include the tcpdump file. I am re-posting just in case this message was over looked for lack of information. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. My server name is vadev2 (Kamailio run on this server), and it's IP Address and port are 172.21.1.108:5060; Asterisk servers are (172.21.1.97:5060, and 172.21.1.124:5060). The rtpproxy is 172.21.1.108:17722 Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 vadev2/usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.21.1.108:59794<http://172.1.1.13:59794> failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work, other than I do not believe anything is going through rtpproxy; I can make calls and use sip tools which pass requests through Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what and why. Should I be using rtpengine instead of rtpproxy? I did attempt rtpengine but it would not compile due to missing json.h files. Thank you for any help -Steve

1 0

MSILO with encryption/decryption
by SamyGo 23 Nov '17

23 Nov '17

Hi, Is there any internal mechanism for msilo to store messages encrypted and retrieve them decrypted before sending? Thanks, Sammy

1 0

RV: CANCEL not being matched by tm module when via address does not have port included
by CARLOS GARCIA MORCHON 23 Nov '17

23 Nov '17

Hi, I have an IMS core deployed using kamailio 5.0.2. Calls from clients are reaching the core pcscf from a freeswitch (with address A.A.A.A) + a kamailio acting as proxy (with address B.B.B.B), so the INVITE request looks like this: INVITE sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Record-Route: <sip:A.A.A.A;lr> Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Via: SIP/2.0/UDP B.B.B.B:5080;received=B.B.B.B;rport=5080;branch=z9hG4bKHpB704ZaXSKtm Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 INVITE Contact: <sip:mod_sofia@B.B.B.B:5080> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 224 P-Charge-Info: <sip:222222222@mobile.tuenti.int>;npi=ISDN;noa=4 X-FS-Support: update_display,send_info Remote-Party-ID: "222222222" <sip:222222222@B.B.B.B>;party=calling;screen=yes;privacy=off If the client cancels the request, the CANCEL request that gets to the core pcscf looks like this: CANCEL sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 CANCEL Content-Length: 0 Reason: Q.850;cause=16;text="Normal Call Clearing" The processing of the CANCEL at the pcscf is eventually reaching the t_check_trans() function, but the logs return “no CANCEL matching found!” and thus the processing ends. I have followed the code up to the via_matching function in the t_lookup.c file of the tm module source and added some logs. The via matching is failing at the port comparison (the port stored for the INVITE is “5060”, the port stored for the CANCEL is “0”. The flow works ok with a different client (connected directly to the core) which includes the port in the Via headers. ¿Can this be a bug in kamailio or am I doing something wrong?. AFAIK The RFC 3261 does not mandate to include the ports in the sent-by element of the Via header Thanks! Carlos ________________________________ Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it. Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

3 4

Kamailio new install, unable to create users
by Atux Atux 23 Nov '17

23 Nov '17

Hi. New to the area and trying to find my way around with kamailio. In one of my debian servers i need to have kamailio with a gui to start playing around with it. i am following the http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour to install kamailio. When running kamdbctl create it asks what format coding that i must have to configure the DB. i simply add latin1 and it goes on. then when i issue kamctl add daniel 1234qwet it comes with the following error: root@debian:~# kamctl add daniel 1234qwet MySQL password for user 'kamailio@localhost': ERROR 1045 (28000): Access denied for user 'kamailio'@'localhost' (using password: YES) ERROR: introducing the new user 'daniel' to the database failed root@debian:~# Any ideas on how to proceed with the following error please?

2 3

kemi with lua and dispatcher - attempt to index field 'dispatcher' (a nil value)
by Karsten Horsmann 23 Nov '17

23 Nov '17

Hello Daniel, hello List, after reading the fresh released kemi documentation (Thanks for that!) on github i start to configure an kemi lua configuration. It seems to be that there is not an dispatcher export to kemi in my installation. Or i missconfigured something? How to use dispatcher with kemi? 1.) First try to solve this - use param "register" "modulename" to app_lua - doesnt work 2.) grep around in the auto kemi in/export messages in debug startup - nothing to see. 3.) grep around in the kamcmd api.list command for lua - no dispatcher here the intressting parts of the configs and lua-script: kamailio.cfg ... loadmodule "dispatcher.so" loadmodule "permissions.so" loadmodule "avpops.so" #!ifdef WITH_CFGLUA loadmodule "app_lua.so" modparam("app_lua", "reload", 1) modparam("app_lua", "register", "dispatcher") #!endif #!ifdef WITH_CFGLUA modparam("app_lua", "load", "/etc/kamailio/lua/kamailio-kemi-lua.lua") cfgengine "lua" #!else import_file "request-route.cfg" #!endif ... kamailio-kemi-lua.lua function ksr_request_route() ... -- dispatch destinations if KSR.permissions.allow_source_address(FLT_EDGE) then KSR.info("new $fU -> $rU in permissions"); direction = "inbound"; routing_target = FLT_FSNODE; ksr_route_dispatch(direction, routing_target); KSR.x.exit(); end KSR.info("NOT-OKAY - $rU dropped"); -- End of Route() KSR.sl.sl_send_reply(407, "Proxy Authentication Required. See IP GUI"); return 1; ... end --- dispatcher route function ksr_route_dispatch(direction, routing_target) KSR.xlog.xinfo("new $fU -> $rU in routing_target ksr_route_dispatch"); KSR.info(routing_target); if (routing_target == nil) then KSR.xlog.xinfo("routing_target nil"); KSR.sl.sl_send_reply(604, "Does Not Exist Anywhere"); KSR.x.exit(); end if KSR.dispatcher.ds_select_dst(routing_target, 4) then send_reply(404, "No destination"); KSR.x.exit(); end ksr_route_myheader(); ksr_route_relay(); KSR.x.exit(); end /usr/sbin/kamailio[26763]: ERROR: app_lua [app_lua_api.c:713]: app_lua_run_ex(): error from Lua: /etc/kamailio/lua/kamailio-kemi-lua.lua:131: attempt to index field 'dispatcher' (a nil value) kamailio -v version: kamailio 5.0.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_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: unknown compiled on 20:20:24 Nov 16 2017 with gcc 4.8.5 rpm -aq | grep kam kamailio-mysql-5.0.4-26.el7.centos.x86_64 kamailio-5.0.4-26.el7.centos.x86_64 kamailio-lua-5.0.4-26.el7.centos.x86_64 kamailio-ims-5.0.4-26.el7.centos.x86_64 rpm -ql kamailio-5.0.4 | grep disp /usr/lib64/kamailio/modules/dispatcher.so /usr/share/doc/kamailio/modules/README.dispatcher /usr/share/kamailio/dbtext/kamailio/dispatcher -- Kind Regards *Karsten Horsmann*

2 5

"Federated" PBX solution with GUI to use with Kamailio - Suggestions needed
by George Diamantopoulos 23 Nov '17

23 Nov '17

Hello all! I'm looking at "PBX" platforms to use with kamailio as a load-balancer. This is not a strictly kamailio question, but I thought that if anyone has a suggestion to make, they would be watching this list. I'm investigating a use case with the following requirements: * Kamailio will act as a load-balancer, dispatching initial requests to PBX entities (e.g. Asterisk, Freeswitch) * PBX solution must have a single GUI interface to manage PBX functions (think FreePBX, FusionPBX etc) * Configuration via the GUI must apply to all PBX instances * Some sort of "asterisk-realtime + device state synchronization"-like setup must ensure every PBX box can service any request at any time (for failover) * Solution needn't be multi-tenant oriented, but if it is, it must work fine for single-tenant setups as well without multi-tenant capability getting in the way all the time * Feature list needn't be as full as, say, FreePBX's, but most common functionality must be available (inbound/outbound routing, trunks, queues, hunting to name a few) * Must be open source As a proprietary example, MiRTA PBX seems to check all the boxes above, apart from being open source. Now this "product" also offers high availability for databases and web apps, which is beyond the scope of this investigation. I'm only looking at a solution that provides a federated, synchronized farm of PBX instances. Any input would be greatly appreciated. Thanks! Best regards, George

3 2

Re: [SR-Users] Kamailio IMS + Asterisk PBX
by Amar Tinawi 22 Nov '17

22 Nov '17

My tests results : Unfortunately, no scenario was success, maybe because Asterisk does not support sh interface (Diameter) My test was for presence service and couldn’t get it done. The results of Having asterisk as an AS with kamailio IMS : with sip trunk configured towards S-cscf And whether the user is identified in Asterisk as extension user or not asterisk responded with 404 user not found. Sent from Mail for Windows 10

2 1

debian setup without registration
by Atux Atux 22 Nov '17

22 Nov '17

Hi. i am new to the are of kamailio. i am looking to install in a PC running debian 9 kamailio. that is easy since all i had to do was apt-get install kamailio. I am looking on how to setup the system in a way that everyone sending a register, to be able to register no matter what user name or passwd sends. i do not want a MYSQL server on that server. Could someone give me some hints on how to configure it please? In /etc/kamailio/kamctrlc i have SIP_DOMAIN=192.168.124.5 which is my IP. Everybody from my LAN 192.168.124.0/24 network must be able to send a register with any username and any passwd and be able to register. Also from the outside world, through port forward from my router. Some hints please?

3 2

kamcmd gives segmentation fault (5.0.4)
by José Lopes 22 Nov '17

22 Nov '17

Hello, I invoke kamcmd inside Freeswitch Lua Scripts and right now i am getting segmentation fault any time. This started to happen after I upgraded kamailio from 4.4 to 5.0.4 . I tried to analyse the core dump and it give me the next information. When I execute on the console, the command works. Am I doing something wrong? There is any alternative or better way to communicate with kamailio? Thanks in advance Best regards, Jose Lopes Core was generated by `/usr/sbin/kamcmd -s udp:127.0.0.1:2050 uac.reg_refresh s:29010017'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 591 kamcmd.c: No such file or directory. (gdb) bt #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 #1 0x0000000000403eee in run_binrpc_cmd (s=3, cmd=0x7ffec898a8d0, fmt=0x0) at kamcmd.c:924 #2 0x0000000000404499 in run_cmd (s=3, cmd=0x7ffec898a8d0, format=0x0) at kamcmd.c:1068 #3 0x00000000004073aa in main (argc=5, argv=0x7ffec898d1e8) at kamcmd.c:2114

2 2

Kamailio IMS + Asterisk PBX
by Amar Tinawi 21 Nov '17

21 Nov '17

Dears; Good day ! is it possible to use Asterisk pbx server as an Application server for the kamailio IMS users ? as if i want to correlate an IMS user with an extension number. My though is kamailio ims will handle the registration process and authentication and Asterisk will deal with IMS users as they are PBX users to get all asterisk's benefits (call conference, monitoring, waiting, voice mail ... etc) i'm thinking to integrate Asterisk with S-cscf, and when an IMS user register to the IMS to domain, a trigger will initiate the S-cscf to send notify to asterisk . i didnt find any solution or tutorial for this procedure, and idont know if it is possible to be done. Regards Ammar

5 8

Adding strip and preix attribute to Dispatcher table
by Ashutosh chaubey 21 Nov '17

21 Nov '17

Hi, Is there a way to add prefix and strip in dispatcher table attribute column. I have tried prefix=11,strip=3 but this seems to not work. Any help? Regards, Ashutosh Chaubey

2 4

http_async_client problem with tls_ca_path
by gmele 20 Nov '17

20 Nov '17

Hello, I'm having strange behavior with the http_async_client module and https connections. Our kamailio config use REST interfaces to send push to mobile apps when calls are initiated. We also use a REST interface to register the client in our push database (and then be able to use the info to send a push). Our REST interface use TLS to cypher exchanged data between the SIP registrar and the Pusher service. We have configured the tls connection using modparam: # ------ http async client params ------- modparam("http_async_client", "workers", 16) modparam("http_async_client", "connection_timeout", 20000) # 20 seconds timeout on HTTP requests modparam("http_async_client", "hash_size", 2048) # update this value depending of the load #modparam("http_async_client", "curl_verbose", 1) # curl debug modparam("http_async_client", "tls_ca_path", "/etc/kamailio/ssl/ca") What we have noticed is when there is 1 SIP Register requests triggering and HTTPS request all works fine, but as soon as we have several HTTPS request in //, the HTTPS request fails with curl error 77: curl: (77) Problem with the SSL CA cert (path? access rights?). We tried to put the Root CA Path when building the request: $http_req(all) = $null; # reset the parameters $http_req(method) = "POST"; $http_req(hdr) = "authorization: "+$hdr(authorization); $http_req(hdr) = "Content-Type: application/json"; $http_req(suspend) = 0; # don't suspend the transaction, continue routing script's execution $http_req(body) = "{ 'pushRegistration':{'callId':'" + $ci + "', 'fromUri':'" + $fU + "@" + $fd + "', 'contactLine':'" + $ct + "'}}"; $http_req(tls_ca_path) = "/etc/kamailio/ssl/ca/"; http_async_query("https://pusher:443/rest/push/register", "HTTP_REGISTER_REPLY"); But we have exactly the same behavior. Does the http_async_client module supports muti-thread? Regards Giovanni -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 17

Re: [SR-Users] 183 acc records even if early_media equals to 0
by Daniel-Constantin Mierla 20 Nov '17

20 Nov '17

Hello, can you run list command in gdb to see the code? Because I am not sure what is exactly at that line in your version with the patch. Cheers, Daniel On 09.11.17 18:10, Marco Capetta wrote: > I applied the patch but kamailio crashes with the following error: > > (gdb) bt full > #0 0x00007f8af1e9b9ae in relay_reply (t=t@entry=0x7f8a29c05868, > p_msg=p_msg@entry=0xffffffffffffffff, branch=branch@entry=0, > msg_status=<optimized out>, > cancel_data=cancel_data@entry=0x7ffd92443570, > do_put_on_wait=do_put_on_wait@entry=0) at t_reply.c:1968 > relay = 0 > [...] -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 7

Dmq usage
by Aidar Kamalov 20 Nov '17

20 Nov '17

Hello, I want to distribute some messages-actions for redis to my kamailio servers. I use this code: For send: dmq_bcast_message("redis", "set early:$fU", "text/plain"); For recieve: request_route { xnotice("ROOT: $rm $rU [$ci]"); if(is_method("KDMQ")) { if($rU == 'redis') { route(REDIS); } else { dmq_handle_message(); } } ...... route[REDIS] { xnotice("REDIS: $rb"); redis_cmd("redis", "$rb", "r"); return; } When dmq_bcast_message starts - other node recieve 7 KDMQ sip packets. And not always it executed. Is it right way to use dmq? Dmq user locations are ok.

2 8

kamailio-wiki dead links to github.com/....master branch
by Karsten Horsmann 20 Nov '17

20 Nov '17

Hello, https://www.kamailio.org/wiki/devel/config-engines has links to https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… for example configs. All the "kemi" stuff is gone. 404 Error. Where i found actuall (kemi / kamailio 5.0.x) samples? -- Kind Regards *Karsten Horsmann*

3 5

no event received for call disconnection
by Aleksandar Sosic 20 Nov '17

20 Nov '17

Hi Everyone, we're trying to implement cgrates in our kamailio nodes. Everything is working fine except for the fact that kamailio is not signaling the call end to cgrates for prepaid users... The versions, logs and confs are: ``` # kamailio -v version: kamailio 5.0.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_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: unknown compiled with gcc 5.3.1 ``` ``` # cgr-engine --version CGRateS 0.9.1~rc8 git+15a0793 (2017-10-23T12:15:40+02:00) ``` The kamailio conf regarding our problem is: ``` # Inform CGRateS about CALL_END (stop debit loops, perform accounting if desired in this way) route[CGR_CALL_END] { if $sht(cgrconn=>cgr) == $null { xlog("Charging controller unreachable"); exit; } $var(callDur) = $TS - $dlg(start_ts); evapi_relay("{\"event\":\"CGR_CALL_END\", \"callid\":\"$dlg(callid)\", \"from_tag\":\"$dlg(from_tag)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_answertime\":\"$dlg(start_ts)\", \"cgr_duration\":\"$var(callDur)\", \"cgr_supplier\":\"$dlg_var(cgrSupplier)\", \"cgr_disconnectcause\":\"$T_reply_code\"}"); } ``` The kamailio logs: ``` 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} ``` Thanks for your help, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

2 13

kamailio 4.4 - jansson_get_field parse error actions
by Jonathan Hunter 17 Nov '17

17 Nov '17

Hi Guys, I am looking to send calls to a default route should we have issues when connecting to our API which provides routing instructions. On scenario is handling an invalid JSON response , which would result in a parse error. Is it possible to capture when there is a failed to parse error when dealing with JSON responses to an http query in kamailio? In particular when using the jansson_get_field, I am looking to default to a route should their be a parse error, so I am forcing an erro, I get this in the logs, but can I grab to act on it? Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' On receipt of this I would then like to hit a default route. Many thanks Jon

1 1

SUBSCRIBE NAT Keep-Alive
by Asgaroth 17 Nov '17

17 Nov '17

Hi All, I'd like to ask the guru's out there their opinion on how to solve the nat problem for presence subscriptions. We have an issue with subscriptions that get timed out and the notify clears the subscription out of the active_watchers table due to the notify timing out, I think this is expected and the right way to do it. However, how do others out there ensure that their subscription nat states are kept alive? For registrations we have nathelper, but, what is the sanest way of doing the same for subscriptions? I was originally thinking of sending notifications through our registrar/lookup server, but, I'm not sure if that is the correct way to do it. Any thoughts, tips, tricks would be much appreciated. Cheers

3 3

Kamailio questions
by Wilkins, Steve 17 Nov '17

17 Nov '17

Hello, I am new to Kamailio and to the Kamailio Mailing list and hope I am using it appropriately. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. For the sake of the argument, my server name is myServer, and it's IP Address is 172.1.1.13; Asterisk is on a different server. Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 myServer /usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.1.1.13:59794 failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work; I can make calls and use sip tools which pass requests through the Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what at such a high port. Question: The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Note: Even with no rtpproxy, I get the same error mentioned above in the Issue section. Thank you,

2 4

dr_load_routing_info(): failed to add rule id 4 -> skipping
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i experiment with the drouting module with mysql-db. I try to route with "alphanumeric" stuff - like 'tetris' in sip://tetris@mytest.example https://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp31352844 The DB-Field is an 64 string like route-id. In dr_rules.route-id i can use "named-routes" if i leave the prefix empty or i fill in only numbers. With strings i got this error: ERROR: drouting [dr_load.c:555]: dr_load_routing_info(): failed to add rule id 4 -> skipping SQL Result for rule-id 4: # dr_rules # ruleid, groupid, prefix, timerec, priority, routeid, gwlist, description '4', '9000', 'tetris', '', '0', 'tetris', '4', 'testroute' Is drouting only for "number-routing"? -- Kind Regards *Karsten Horsmann*

3 6

best practice to dynamic route block based on $rd - Domain in R-URI
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i am now at the point that one kamailio is not enough :). So i decide to set another kamailio internal as sip-router based on drouting. My config is straight forward the example config with additional drouting stuff. Whats the best practice to route based on $rd with kamailio? I dont want to end with an bunch of if then else. Should be come from database. Pseudo Code: if ($rd == old.sbc.domain) { route[oldstuff]; } else { route[newstuff]; } Is there an module that i can use? -- Kind Regards *Karsten Horsmann*

1 1

How to use $var as arguments to module functions
by Duarte Rocha 16 Nov '17

16 Nov '17

Greetings, Is there a way to use the NATHELPER functions with a variable insted of a string with the value ? For example, I can do "fix_nated_sdp("11");" but Kamailio throws an error if i use "fix_nated_sdp("$(var(nat_fix)");" or any other variations. How should i proceed to be able to do this ? Thanks

2 1

play music while processing a http request
by 赵国杰 16 Nov '17

16 Nov '17

Hello guys, I have the following senario: ua1 calls ua2. when the INVITE reaches the kamailio, before foward to ua2, kamailio has to do a http request using “http_async_query”. However, the http request can take a really long time. I want the kamailio start playing music when the http request is sent and stop the music when the http response is received. My config file is like this: request_route { ... t_newtran(); http_async_query("url here", "HTTP_REPLY"); } route[HTTP_REPLY] { ... # forward the invite to ua2 } how do i complete the config? Thanks

2 3

Dmq_usrloc
by sofos koko 14 Nov '17

14 Nov '17

Hi ...i would like to connect two kamailio in diffwrent network so i can get call from one to another is any kamailio.cfg example to help me do it!

1 0

topos and topos_redis : error log -> ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store
by Abdoul Osséni 14 Nov '17

14 Nov '17

hello, After activated redis backend for topos module, I have some error logs. Can you help me to understand these logs? # kamailio -V version: kamailio 5.1.0-rc0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_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: unknown compiled on 14:18:08 Nov 14 2017 with gcc 4.9.2 Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Abdoul OSSENI

1 0

AVPOPS: is_avp_set/avp_check "name" parameter as variable.
by Asgaroth 14 Nov '17

14 Nov '17

Hi All, I was having a read of the AVP module documentation and I see mentioned that in future releases of kamailio that the AVP module may become deprecated in favour of using AVPOPS module. To this end, I started looking to switch some of AVP functions over to AVPOPS functions when I cam across an issue where I cannot have the name parameter of the avp to check a variable. I have some attributes defined for a domain which I load up using lookup_domain, one of these attributes is a dispatcher setid based on the request method being processed. So, with AVP I can do something like the following to see if it is set: if ( attr_exists( $_s(dm_ds_setid_$rm) ) ) The above works as expected, but if I try using something equivelent using avpops functions, something like: if ( is_avp_set("$avp(s:$_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(s:dm_ds_setid_$rm)") ) or if ( is_avp_set("$avp($_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(dm_ds_setid_$rm)") ) Then it fails the check even though the avp is set correctly, there are no errors in the logs so it all appears to be okay. Likewise, when checking with the following AVP function, it works: if ( attr_equals("$_s(dm_ds_setid_$rm)", "100") ) But, if I try with AVPOPS function, it fails (I'm guessing its bacause the avp cannot be found) if ( avp_check("$avp(s:$_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(s:dm_ds_setid_$rm)", "100") ) or if ( avp_check("$avp($_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(dm_ds_setid_$rm", "100") ) Are the AVPOPS functions able to take nested variables like the above? Is this a bug or am I using them incorrectly? I'm currently trying this out on kamailio v5.0.4. Any thoughts on the issue would be appreciated. Thanks

2 2

max_contacts does not work as expected
by Gholamreza Sabery 14 Nov '17

14 Nov '17

I have a Kamailio server that has: modparam("registrar", "max_contacts", 2) But recently I noticed some users have more than two records in the location table. For example one of the users has three different records with nearly the same information. The records are as follows. Is this behavior normal? (I am using Kamailio 4XX). *************************** 1. row *************************** id: 171296745 ruid: uloc-2c69-59db6c73-7ca-132d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: XpCnmaAhRF cseq: 43 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 14 *************************** 2. row *************************** id: 171296746 ruid: uloc-2c69-59db6c73-7f2-eb5d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: BzMKuyJN8C cseq: 50 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 23 *************************** 3. row *************************** id: 171296747 ruid: uloc-2c69-59db6c73-80f-541d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: 3p9P4PgZSv cseq: 51 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 15

1 0

Load Gateway according hostname
by Ashutosh chaubey 13 Nov '17

13 Nov '17

Hi, I am trying to find a way to load a specific gateway but instead of lcr_id i want it load gateways according to the hostname. I am passing Hostname as a header to kamailio so its easy to access that but unabale to load gateways according to it. Any help? Regards, Ashutosh Chaubey

1 0

using bcrypt passwd hashing
by Walter Martín Villalba 13 Nov '17

13 Nov '17

Hello, I did some searches online and talked to some colleagues and it seems Kamailio only supports the traditional HTTP digest authentication, which uses MD5. I would like to know if any of you has been successful in using bcrypt/scrypt/pbkdf2 passwd hashing, instead of MD5, which has been deemed as obsolete and insecure a long time ago. Perhaps you've written your own auth module, or just modified the config script to call some other credential checking routine using a custom python/perl script (I'm thinking of doing the latter, of nothing better is available). If any of you have done something like this, using bcrypt or any other current and secure hashing algorithm, I would appreciate some guidance. If you haven't, aren't you concerned about storing MD5 password hashes in your database? Note: if I can't find a good answer using this list, I will try the developer's list next. Thanks in advance, Martín.

4 5

Kamailio issue
by bazou2 bazou2 13 Nov '17

13 Nov '17

Hello, I have a kamailio version 4.4 with an Asterisk 13 ... and i need to make a call from a sip phone app.(i have a user subscribed on the kamailio to be used on the sip phone app). The weird thing that first time, the call works (I can see the sip messages working fine between the app kamailio and asterisk) but when i try again it doesn't work. SIP messages show that the kamailio and Asterisk keep send trying request between them and the loop of these kind of messages doesn't end.. I need to know if i am missing something in the configuration file of the kamailio or what? Regards Sent from Outlook<http://aka.ms/weboutlook>

2 2

Development open in master branch (to be v5.2.x)
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, git branch 5.1 was just created (to host the release series v5.1.x), therefore new features can now be pushed again in master branch. They will be part of the next future release, likely to be numbered 5.2.x. Any fixes that affect existing code in branches 5.1 or older version have to be backported - push first to master and then cherry pick -- see the contributing guidelines at: - https://www.kamailio.org/wiki/devel/git-commit-guidelines#backporting_commi… Many thanks to all contributors so far! Testing of branch 5.1 and giving feedback for it is very appreciated! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Branch 5.1 created
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, the GIT branch 5.1 has just been created, it will host the release series 5.1.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 5.1 origin/5.1 Hopefully in two-three weeks time frame the full release of 5.1.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 5.1. No new features can get in branch 5.1. Enhancements to documentation or helping tools, as well as kemi exports are still allowed. If you are not sure about doing or not a backport, ask on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

GeoIP2 module deb package
by Joel Serrano 12 Nov '17

12 Nov '17

Hi, I can't seem to find the geoip2 module for debian stretch (installing from packages, kamailio 5.0 branch). Is it not available in the repository? Thanks, Joel.

2 4

Kamilio for IMS and VoLTE Setup Questions
by Jack Folgolburg 11 Nov '17

11 Nov '17

Hello, I saw the Kamailio World 2017 presentation "Kamilio for IMS and VoLTE" (https://www.youtube.com/watch?v=Wdhcdv5JVPw). I would like to install the servers for evaluation and testing. The slides say to use Ubuntu 16.04, but there are no xenial packages at the ng-voice repository. Is it okay to install on Ubuntu 14.04, since there are trusty packages? I have done this and got through all the configuration prompts. After installation, is there any way to confirm that the site is operational? How to bring up the management console (i.e. what is local URL)? In the presentation, it looks like a commercial site at ng-voice is accessed. I understand that there is a REST API. Is this API documented anywhere? Maybe, the most important question is, after installation, what is the fastest way to add two users and test access from IMS clients? Thank you, James

1 0

183 acc records even if early_media equals to 0
by Marco Capetta 09 Nov '17

09 Nov '17

Dear All, I'm facing a strange problem with the call accounting module: even if in my configuration I have the parameter: modparam("acc", "early_media", 0) I can find some ACC records with sip_code 180 or 183. I investigated those cases and this issue seems to happen when an endpoint sends an 183 message immediately followed by a 200 OK. The issue seems started from kamailio version 4.4.1, but I cannot be really sure of this. Have you ever had a similar problem? Do you have any suggestions on how to solve it? Thanks Marco

2 11

t_relay_to_udp - 500 server error occurred (6/SL)
by Donat Zenichev 09 Nov '17

09 Nov '17

Hi all. I have a problem with t_relay_to_udp function. My topology: UAC -> tcp -> kamailio <-> udp <-> asterisk Kamailio processes requests from clients over TCP and relays requests to asterisk over UDP. Kamailio gets INVITE from client, relays it to asterisk, then he sends 100 trying to user client and abruptly sends 500 internal error to client (after 100 trying). But the main interesting thing is, that asterisk answers on INVITE and dialog continues between kamailio and asterisk (without user agent client) - it reaches the destination then, without originator : ) How error on kamailio looks like: Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: tm [t_fwd.c:1716]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (6/SL) The same problem is, when I change transport protocol between kamailio and asterisk to TCP, using the t_relay_to_tcp function. When I use t_relay function, it works fine, nothing happens that can drop dialogs between kamailio and clients. Relaying code part: if ($var(to_asterisk)=="1") { xlog("L_NOTICE","Relaying to asterisk - change socket to udp:10.1.1.1:5070 \n"); force_send_socket(udp:10.1.1.1:5070); if (!t_relay_to_udp()) { sl_reply_error(); } } I need to use t_relay_to_udp with force_send_socket, because kamailio changes the source port (source socket) every time he generate new INVITEs to asterisk and this is irrelevant. Dump between kamailio and user agent client: INVITE sip:1111111111example@kam1.domain.com:5070;transport=TCP SIP/2.0 Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z- Max-Forwards: 70 Contact: <sip:debug.device-example@uac.domain.example:5065;transport=TCP> To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="debug.device-example",realm=" kam1.domain.com ",nonce="WgQGlVoEBWnhH3CF6DE77uKxKJx0tT4e",uri="sip:1111111111example@kam1.domain.com:5070 ;transport=TCP",response="4753aa849715d26d0dde1b3c2314c843",a orithm=MD5 Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 v=0 o=Z 0 0 IN IP4 uac.domain.example s=Z c=IN IP4 uac.domain.example t=0 0 m=audio 8000 RTP/AVP 8 3 110 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: MS Lync Content-Length: 0 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL) Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070 ;transport=TCP>;tag=9a78ee8ed9a3d4ba441c41d83ba2e904.4113 From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: example Content-Length: 0 Thanks in advance for hints. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-800 http://wnet.ua

1 0

Topoh module configuration for SIP proxy and SIP registrar
by Christian Conus 08 Nov '17

08 Nov '17

Hello, I am trying to configure the topoh module for kamailio 5.0.2. I read the documentation but some points are still not clear to me. The topology of my network is as follow: - Two SIP proxies - Two SIP registrars - Only one SIP proxy is running (the other take over automatically in case the first fails) - Both registrars are running (proxy sends requests to both registrars using round robin algorithm) Text mode graph of the topology (hope it helps understand): I tried to enable topoh on SIP proxy only, on both proxy and registrars, with same mask_key everywhere and with different mask_key between proxy and registrars but everytime I could still see parts of the SIP header which was not obfuscated. My questions are: - Is it necessary to enable topoh module on each SIP proxy and SIP registrar ? - The mask_key should be the same on each server or it should be different between proxy and registrars? - the mask address should be the same on each server or it should be different between proxy and registrars ? Thanks in advance for your help, Christian -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 10

evapi_async_relay() failed
by Abdul Basit 07 Nov '17

07 Nov '17

Hi list, I am configuring CGRateS with kamailio 4.4 using deb packages on debian 8. I am facing a issue while processing event_route[dialog:start] for answering call INVITE at evapi_async_relay("{\"event\":\"CGR_CALL_START\", with following error /usr/sbin/kamailio[13887]: ERROR: evapi [evapi_mod.c:288]: w_evapi_async_relay(): failed to suspend request processing However, after discussion with Dan on IRC i replaced function evapi_async_relay() with evapi_relay() resolved the issue. Whats causing this error? Can anyone suggest some way out for keep using evapi_async_relay? -- regards, abdul basit

2 1

Kamailio SIP-I
by Saleem Manasfi 07 Nov '17

07 Nov '17

Dears, Please if you can let us know how does Kamailio reads the SIP-I body (ISUP)? Is there any document or link that may help?

2 1

Re: [SR-Users] Regarding Opensource DID Routing Solution using Kamailio
by surendra tiwari 06 Nov '17

06 Nov '17

Hi All, I have used the kamailio and redis to create the DID routing solution.please find below in github.this repository is created to help the person who want to setup there own DID routing solution with following feature: 1. Inbound Termination with Carrier IP validation. 2. Carrier LCR for DID/TFN to PSTN forwarding. 3. Inbound Abuse Block. 4. CDR in MongoDB. 5. IPTables Block for Sip-Scanners. 6. Integration with RtpEngine. 7. RedisDB for quick DB Access. https://github.com/surendratiwari3/KamInboundSIP On Mon, Nov 6, 2017 at 9:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > your email was addressed to kamailio-announce mailing list (I took your > message from its bounces filter), but that list is only for announcement > issued by Kamailio project (major news about our releases, security > incidents, events, etc...). > > News from community members are welcome and can be sent to: > > - sr-users(a)lists.kamailio.org for non-commercial announcements > > - business(a)lists.kamailio.org for announcements related to commercial > offerings/products (but here you can also send non-commercial announcements) > > Your announcement is not commercial and can be sent to sr-users. Once it > will be there, I will also make a news article on kamailio.org website. > > I didn't have the time yet to check in detail what you put together, but > sounds interesting. > > Thanks for sharing, > Daniel > > (cc-ed Fred as he is moderator for kamailio announce mailing list) > > On 01.11.17 18:46, surendra tiwari wrote: > > Hi All, > > I have used the kamailio and redis to create the DID routing > solution.please find below in github.this repository is created to help the > person who want to setup there own DID routing solution with following > feature: > > > 1. Inbound Termination with Carrier IP validation. > 2. Carrier LCR for DID/TFN to PSTN forwarding. > 3. Inbound Abuse Block. > 4. CDR in MongoDB. > 5. IPTables Block for Sip-Scanners. > 6. Integration with RtpEngine. > 7. RedisDB for quick DB Access. > > > https://github.com/surendratiwari3/KamInboundSIP > > -- > With Regards, > Surendra Tiwari > VoIP Application Developer, > +91-9967609476 <+91%2099676%2009476> > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > -- With Regards, Surendra Tiwari VoIP Application Developer, Plivo Comm +91-9967609476

1 0

Sipcapture HEP Values
by Soltanici Ilie 06 Nov '17

06 Nov '17

Hi all, How can i get the Destination IP from HEP packet? I tried $hep(dst _ip) like in documentation (https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#sipcapture_mo…) - but is not working. (pvar "hep" has an invalid name param [dst_ip]). Thank You.

1 0

Invalid JSON when dumping htable with JSONRPC-S
by Marrold 03 Nov '17

03 Nov '17

Hi All, I am currently trying to retrieve the contents of a htable using the JSONRPC-S module over HTTP, however I have observed if a slot contains multiple values, the JSON returned has duplicate keys which could be considered invalid. Here are the relevant lines from kamailio.cfg - event_route[xhttp:request] { if ($hu =~ "^/carrier") { jsonrpc_exec('{"jsonrpc": "2.0", "method": "htable.dump", "params" : {"htable": "carrier"}, "id": 1}'); xhttp_reply("200", "OK", "text/html", "$jsonrpl(body)"); } } And the output, showing a slot with two items- { "entry": 5, "size": 2, "slot": { "item": { "name": "carrierA", "value": "90", "type": "str" }, "item": { "name": "carrierB", "value": "190", "type": "str" } } } And the same from `kamcmd -s unix:/tmp/kamailio_ctl htable.dump carrier` - { entry: 5 size: 2 slot: { item: { name: carrierA value: 90 type: str } item: { name: carrierB value: 190 type: str } } } Would it be possible to change it to something like the following - { "entry": 5, "size": 2, "slot": [ { "name": "carrierA", "value": "90", "type": "str" }, { "name": "carrierB", "value": "190", "type": "str" } ] } Any other suggestions or work arounds would be appreciated, Thanks Matthew

2 1

Contacts in 302
by Diego Nadares 03 Nov '17

03 Nov '17

Hi Guys, I'm generating a 302 reply from kamailio. In this 302 I append new branches with new contacts. if($var(routing)=~"redirect"){ jansson_get("contacts_len", "$var(evmsg)", "$var(contacts_len)"); xlog("L_INFO", "Contacts len $var(contacts_len)"); $var(i) = 0; while ($var(i) < $var(contacts_len)){ jansson_get("contacts[$var(i)]", "$var(evmsg)", "$var(contact)"); append_branch($var(contact), "0.5"); $var(i) = $var(i) + 1; } send_reply("302", "Moved Temporarily"); exit; } The problem i'm facing is that I can't delete the original contact *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>* The resulting conctact: Contact: *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>*, <sip:1111@188.111.111.112>;q=0.5, <sip:1111@188.111.111.112>;q=0.5 I all ready tried with remove_hf('Contact') and adding the new ones after that but it doesn't worke either. The problem is that one gw takes the first contact over and over again and never the other two. Thanks in advance. Diego.

2 4

Multiple IPs
by Oz Mortimer 03 Nov '17

03 Nov '17

Hi all, I have Kamailio set as an SBC with freeswitch behind - that all works fine. Freeswitch can send calls out VIA Kamailio to external IP address’, but if I try sending a call to another IP hosted by Kamailio the seems to go nowhere… Supplier -> Kamailio -> Customer1 -> Kamailio -> Customer2 The Kamailio server has multiple assigned IPs, which are used to multi-tennant - but I cant for the like of me to get Tennant to tenant calls to work (via freeswitch) Any pointers?

2 13

onreply_route block not processing replies
by José Seabra 03 Nov '17

03 Nov '17

Hi there, I'm facing an issue regarding with replies coming to kamailio that aren't processes inside of onreply_route block. Anyone here can help me understand why these replies (1XX) aren't entering on onreply_route bock? is there any situation already identified on Kamailio that can originate this behavior? The Kamailio version used is 5.0.1. Thank you Regards -- José Seabra

2 3

Branching Kamailio v5.1
by Daniel-Constantin Mierla 03 Nov '17

03 Nov '17

Hello, I am planning to create the got branch 5.1 (to be used for v5.1.x series) by end of next week (likely on Friday, Nov 10, 2017). After that, the master will be open for new features and fixes will need to be backported to 5.1. Should anyone want a different schedule, reply to the mailing lists and propose your plans. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

WebRTC error code 488 in iOS application
by Denys Pozniak 03 Nov '17

03 Nov '17

Hello! Probably this case is not connected Kamailio directly, but it can be somebody will point me in the correct direction :) We have VoIP solution: FreeSWITCH -> Kamailio + Rtpengine -> WebRTC Chrome. Everything works Ok except iOS application based on React Native WebRTC ( https://github.com/oney/react-native-webrtc) + JsSIP. It answers error 488 on incoming INVITE (wSDP): INVITE SDP to iOS: *......* *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* iOS console log: *11:55:23.670 browser.js:133 JsSIP:WebSocketInterface send() +10ms* *11:55:23.670 browser.js:133 JsSIP:RTCSession session progress +5ms* *11:55:23.671 browser.js:133 JsSIP:RTCSession emit "progress" +1ms* *11:55:23.671 webrtc.js:108 WebRTC.on progress: local* *11:55:23.679 browser.js:133 JsSIP:RTCSession emit "peerconnection" +8ms* *11:55:23.680 webrtc.js:89 WebRTC.on peerconnection* *11:55:23.680 browser.js:133 JsSIP:RTCSession emit "sdp" +1ms* *11:55:23.689 browser.js:133 JsSIP:Transport send() +20ms**11:55:23.689 browser.js:133 JsSIP:Transport sending message:* *SIP/2.0 488 Not Acceptable Here* *11:55:23.701 debug.js:115 JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:Error: Failed to set remote offer sdp: Failed to create channels.* I found similar issue on thier git https://github.com/oney/react-native-webrtc/issues/293 and main solution is to struct SDP in way like: *a=group:BUNDLE audiom=audio PORT RTP/SAVPF N M Ka=mid:audio* How to modify SDP on Kamailio side (after Rtpengine modification) like below? *v=0* *o=FreeSWITCH 1508701860 1508701861 IN IP4 52.52.52.52* *s=FreeSWITCH* *c=IN IP4 52.52.52.52* *t=0 0* *a=group:BUNDLE audio* *m=audio 30028 RTP/SAVPF 0 8 102* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:102 opus/48000/2* *a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40* *a=ptime:20* *a=sendrecv* *a=rtcp:30028* *a=rtcp-mux* *a=setup:actpass* *a=mid:audio* *a=fingerprint:sha-1 97:1A:E9:FE:D6:65:98:E6:88:EE:D5:9F:20:A3:19:F8:86:E7:E0:E3* *a=ice-ufrag:9QBO8qg2* *a=ice-pwd:PPGPo35xGBAs6RnTdMYVM9I8fx* *a=candidate:KXENzrffZpQvwxdV 1 UDP 2130706431 52.52.52.52 30028 typ host* BR, Denys

1 1

User Registration with location awareness
by Daisy Stevens 03 Nov '17

03 Nov '17

Hi Guys Please forgive me if I seem to asking stupid questions but I am a complete dummy at the moment regarding this software. I would love to be able to set up two kamailio servers with the idea of expansion in the future to handle SIP registrations, however having a two server model how would I get the calls to route to another user that is registered on the other kamailio box? Kind Regards Daisy -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

2 1

msilo Content-Type for picture and video?
by sscc 03 Nov '17

03 Nov '17

Hello kamailio folks i am successfully storing and dumping messages for off line users using msilo content type text/plain. i want to store off-line messages for pictures and videos. my particular queries are 1) is msilo store picture and videos? (i have observed blob in silo database table which is used for pictures. 2) in case it does what will be declared in content type hdr parameter? any help in this regard will highly be appreciated. regards, sscc -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Re: [SR-Users] Reading the P-Assserted-identity header
by Gertjan Wolzak 02 Nov '17

02 Nov '17

Alex and Daniel, Thank you for the feedback, the tip from Daniel helped me pinpoint the problem. And the the one from Alex showed that. The problem was between keyboard and chair, I should have seen that the SBC removed the PAI. Once I changed that, the PAI arrived at Kamailio and manipulation was a breeze. Thanks guys.

1 0

Tcp socket error with active/active kamailio schema
by Donat Zenichev 02 Nov '17

02 Nov '17

Hi guys. I'm trying to realize active/active schema with kamailio. Topology is as following: active/active kamailio -> routing server (asterisk) Kamailio servers work as registrar/location servers and sbc. Asterisk server performs routing functions for calls. We use sip / sips / ws / wss for signaling. Schema built on aws platform and both kamailio have virtual ip address. So when first kamailio crashes, second one assigns ip address of first kamailio. And vice-versa. Both kamailio listens on both shared addresses all the time (with help of sysctl non lolcal bind function). Problem occurs when first kamailio goes down and second kamailio gets ip address of first one. Asterisk tries to send in-dialog requests (for dialogs that were established on first kamailio), request goes to second kamailio and asterisk gets a socket error. And this is proper behaviour I think, because first kamailio established tcp connection with asterisk from certain port, that is not listened on second kamailio. So that, I think that the solution is to use force_send_socket function, but I'm not confident about it. If anyone has ideas, please send your advices. Thanks in advance. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Kamailio not calling each time
by hanoulia salameh 02 Nov '17

02 Nov '17

Hello, I have a problem with kamailio (v4.4), when i try to call from a sip phone through kamailio, the call is interrupted and hang up automatically but after 4s it continues to retry (I can see in the log file Session Progress) and goes through the asterisk (certified/13.13-cert4) and establishes a normal call (It rings at the client side) while it's still disconnected from the sip phone side But when i try again it works fine sometimes. Am I missing something in my configuration file? What can I show in order to help me? Regards

2 2

Kamailio Realtime Integration with Asterisk
by Floimair Florian 02 Nov '17

02 Nov '17

Hi! I found a few tutorials from Daniel on this topic , the latest being https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astd… . Has anyone ever tried this with a more recent Asterisk version (13 or newer) and using chan_pjsip instead of chan_sip? I'd be willing to update the tutorial once I get it running but I would like to avoid going through obstacles that were already handled ;-) My desired setup is 2 Asterisk boxes (which share configuration, useres and registrations via DB) behind a Kamailio as Load-balancer (got that working already). With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 f.floimair(a)commend.com<mailto:f.floimair@commend.com> http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg

1 0

Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 02 Nov '17

02 Nov '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

4 11

Reading the P-Assserted-identity header
by Gertjan Wolzak 01 Nov '17

01 Nov '17

Hello Kamailions, I receive anonymous calls where the PAI has a prefix which I need to strip before forwarding the call to a proxy. I am trying to read the the header with $hdr(P-Asserted-Identity) and I tried the $ai pseudovariable, but I get an empty reply back every time. xlog("PAI is $ai"); results in ; "Nov 1 16:19:55 proxy04 /usr/local/sbin/kamailio[14814]: ERROR: <script>: PAI is <null>" Same for xlog("PAI is $hdr(P-Asserted-Identity). This works for the Privacy header but not the P-Asserted-Identity header. Hope someone can tell me what I am doing wrong.... Rgds, Gertjan

3 2

Record-Route IP Value
by Soltanici Ilie 31 Oct '17

31 Oct '17

Hi All, Having the following Header: Record-Route: <sip:192.168.1.1;lr;did=637.07c7c2d7> How can i extract only IP value from this header? Thank You.

2 3

module version mismatch for /usr/local/lib64/kamailio/modules/kex.so
by Abdul Basit 31 Oct '17

31 Oct '17

Hi, My kamailio installation with 5.0.3 was working fine. I update kamailio source to 5.0.4 and make all; make install I started receiving error below: /etc/init.d/kamailio start [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(7426) ERROR: <core> [core/sr_module.c:431]: version_control(): module version mismatch for /usr/local/lib64/kamailio/modules/kex.so; core: kamailio 5.0.3 (x86_64/lin[FAILmodule: kamailio 5.0.4 (x86_64/linux) ... failed! failed! kamailio.cfg is same as earlier. I removed kamailio sources and modules folder /usr/local/lib64/kamailio/modules/ I also dropped old kamailio postgres database and created new one after fresh source install of version 5.0.4 from git. # git clone https://github.com/kamailio/kamailio kamailio # cd kamailio # git checkout -b 5.0 origin/5.0 But this error persists. Whats the resolution? Is this a bug? -- regards, abdul basit

2 2

Source comparisons by DNS hostname
by Alex Balashov 31 Oct '17

31 Oct '17

Hi, Is there a reasonable way, in Kamailio, to determine whether the source of a given SIP message is equivalent to the IP address to which a given DNS hostname resolves? i.e. if(src_ip == 'fqdn.domain.xyz') but not src_ip, obviously. I can't find any core keywords that would seem to fit the bill for a composite comparison. Any other options? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Radius accounting and cdr's
by Jan Hazenberg 31 Oct '17

31 Oct '17

Hi all, I have a question regarding radius accounting. I would like to have a config where i store local cdr records (and not the ACC records) and send accounting to radius as well. I have the following config: # ----- acc params ----- /* General settings */ modparam("acc", "log_level", 0) modparam("acc", "log_facility", "LOG_LOCAL2") /* Enable cdr's */ modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_LOCAL2") modparam("acc", "cdr_on_failed", 1) /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) /* Accounting triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACCMISSED) modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") This works but stores the raw accounting records as well. If i disable the acc log_flag it stops sending radius accounting records (while the acc_radius log_flag is still enabled). Is it possible to send only radius accounting records and disable the local logging of those records, but leave the cdr logging enabled? Or is there a option where i can set only the cdr_log_level so accounting records will only show up when i increase the debug level? Another option is to use a seperate log facility for accounting and cdr's, but it would be nice if it is possible to disable the logging of accounting records and only keep the cdr's. I'm running kamailio 5.0.3. Thanks, Jan

2 2

Re: [SR-Users] Kamailio for Unified Communication
by Angel Elena 31 Oct '17

31 Oct '17

kamailio is a "proxy" server.... this features may be added by a media-server... freeswitch, asterisk, seems. You can put kamailio as "sip proxy" and you can add these services behind kamailio proxy -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Zarafa Webapp. -----Mensaje original----- De: Jayanth Acharya <jayachar88(a)gmail.com> Enviado: Mar 31-10-2017 10:40 Asunto: [SR-Users] Kamailio for Unified Communication Para: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>; > Hi, > > Can Kamailio be used to provide following broad-type of services ? > > 1. Unified Communication services > 2. Contact Center services > 3. Instant Messaging / Chat services > 4. Voicemail service > 5. Presence aware communication services > 6. Desktop sharing service > > For many/some of the above, I'd expect back2back user-agent capabilities to be > mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are > there some modules that may be used for this purpose ? > > thanks, > Jay > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > >

1 0

Kamailio for Unified Communication
by Jayanth Acharya 31 Oct '17

31 Oct '17

Hi, Can Kamailio be used to provide following broad-type of services ? 1. Unified Communication services 2. Contact Center services 3. Instant Messaging / Chat services 4. Voicemail service 5. Presence aware communication services 6. Desktop sharing service For many/some of the above, I'd expect back2back user-agent capabilities to be mandatory, and if I am not mistaken Kamailio doesn't support B2BUA mode. Or are there some modules that may be used for this purpose ? thanks, Jay

1 0

Reinvite/ACK race
by Alex Balashov 30 Oct '17

30 Oct '17

Hi, I've got a scenario like so: UA A -----> Kamailio P ----> UA B 1. UA A initiates call through Kamailio P; 2. Dialog is established and confirmed, with Record-Route; 3. UA B sends reinvite #1 through P to A; 4. UA B sends 2xx reply; 5. UA B sends end-to-end ACK for reinvite #1 and almost simultaneously sends reinvite #2. The temporal delta is between reinvite #2 and ACK for reinvite #1 on the wire is 3 ms. So, the result — for all kinds of stochastic processing and userspace scheduling type reasons — is that the reinvite is forwarded first, before the ACK. That leads to a 500 / 491 scenario UA A. The cause, as we all know, is that Kamailio's worker threads are loosely coupled, and incoming UDP datagrams are distributed directly by the kernel. Is there any general guidance on what to do with these scenarios? I looked at RFC 5407 § 3.1.4, which appears to describe a similar, but not identical scenario involving an initial INVITE and subsequent reinvite. As far as I can tell, the recommendation in that standard is "space the messaging out more in time". Switching to TCP would presumably help, since any given flow would involve a single connection to a single worker thread and the transport would guarantee ordering. However, that's not really feasible in this implementation for a host of reasons. I know Kamailio has some config locking primitives, but I am extremely wary of complex synchronisation on dialog-identifying attributes, particularly if there is a possibility that such locks could stall a worker perpetually. Any other thoughts welcome! Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

unexpected behavior of save from registrar module
by Vasiliy Ganchev 30 Oct '17

30 Oct '17

Hi, community! I have in my Kamailio script following block (while REGISTER processing, after auth succeed): ..... /* Save to usrloc */ lock("LOCATION_LOCK"); if (!save("location")) { unlock("LOCATION_LOCK"); sl_reply_error(); exit(); } unlock("LOCATION_LOCK"); ...... According to module documentation - if "save" - failed (for any reason) - I expected the control to be returned to the script with error code "-1" But, there are cases - when this does not happen - which lead to deadlock for workers processing REGISTERs (I never unlock("LOCATION_LOCK")): - module answers with internal error - and return 0 from "save" function: https://github.com/kamailio/kamailio/blob/master/src/modules/registrar/save… - line 1013 My use case happened due to malfunctioning device that in some moment of time: - send REGISTER update - kamailio answers 200OK - device retransmit the same REGISTER (same CSeq) - during 20 seconds (hardcoded value in registrar module) - kamailio answers 200OK (and everything is ok) - if retransmission continue more than 20s - kamailio answered 400 Bad request, and did not return control to script (as a result I had no chance to unlock("LOCATION_LOCK") from code above) Tested it on kamailio 4.2.5 and 5.0.3. as for me, possible solutions are: - return -1 instead of 0 from "save" by default - if there are some reasons not to do this - we should change documentation, and explicitly mention that there are cases - when script execution is stopped. thanks in advance! cheers -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

3 5

sanity passes invalid uri
by Juha Heinanen 30 Oct '17

30 Oct '17

I noticed that K 5.0 sanity() test passes uri that contains ` (back quote) character. Config: modparam("sanity", "default_checks", 1024) /* URI checks */ modparam("sanity", "uri_checks", 3) /* RURI, From */ xlog("L_INFO", "Checking $ru\n"); if (!sanity_check()) xlog("L_INFO", "Check failed\n"); else xlog("L_INFO", "Check passed\n"); Syslog: Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Checking sip:jh@te`st.fi Oct 27 17:17:38 lohi /usr/bin/sip-proxy[31946]: INFO: Check passed According to RFC3261: hostname = *( domainlabel "." ) toplabel [ "." ] domainlabel = alphanum / alphanum *( alphanum / "-" ) alphanum Is this a bug or am I missing something? -- Juha

2 2

Remove all cpecial characters (\n,\t etc)from xml body
by Yuriy Gorlichenko 30 Oct '17

30 Oct '17

Hi. I building external presentity service for web based phones Main trouble that i have at the xml body of PUBLISH message \n \t characters and because of it can't set body as string to the redis So I trying to find a way to remove special characters form the xml body for get oneline string instead of multiline I use native .cfg file for now and, unfortunattely, can not use lua or phyton I tried $var(body)=$(rb{s.rm,$var(RemoveThisStaff)}) where $var(RemoveThisStaff) = "\n" and "\t" but it was unsuccesfull. Maybe someone will help to kno how to do it.

2 2

Issue with sipcapture
by Nathan Leyton 28 Oct '17

28 Oct '17

Hi There I am new to Kamailio and have just installed and got it up and running as a sipcapture node. It is working well and logging all the hep sip traffic in my homer database. I have an issue with rtcp extracted to the msg column is not json. It is some kind of weird encoded string (not base64), this is not readable by homer in any way. When I use nonsip_hook I am able to extract the json from the hep and put it in the msg field. However when I return 1 from the hep route on sip to go back to main route I get parsing errors like so on all the sip messages:- Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: sipcapture [sipcapture.c:2711]: nosip_hep_msg(): couldn't parse sip message Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/receive.c:186]: receive_msg(): core parsing of SIP message failed (178.78.120.2:44389/1) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [(null)] Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22518]: WARNING: <core> [core/receive.c:193]: receive_msg(): parsing relevant headers failed Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Oct 27 15:03:41 ip-172-31-19-41 /usr/local/sbin/kamailio[22517]: ERROR: <core> [core/parser/msg_parser.c:675]: parse_msg(): ERROR: parse_msg: message=<HEP3#002,> It never actually enters the main route. With nonsip_hook off, I enter the main route and all my sip messages are there in homer, just missing rtcp as the data is a weird encoded string. Hope someone can help Thanks Nathan -- Nathan Leyton / Chief Technology Officer nathan(a)callhandling.co.uk / 07482 917 412 Call Handling Ltd Office: 0333 321 0888 / Fax: 0333 321 0877 Bakerloo Chambers304 Edgware Road, London, W2 1DY www.callhandling.co.uk [image: htmlsig.com] <https://htmlsig.com/t/0000001BREBFR>

2 1

fix_nated_contact and IPv6
by Sebastian Damm 27 Oct '17

27 Oct '17

Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: <sip:bob@192.168.8.132> outbound Contact: <sip:bob@1234:1234:0:1234:0:0:0:2> The correct way of transforming the Contact URI would be: Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]> This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian

4 4

Too Long Contact URI
by Soltanici Ilie 27 Oct '17

27 Oct '17

Hi All, I cannot Register some softphones because of too long Contact header when using TCP protocol. Using UDP - everything works ok and this is how the Contact header looking: sip:XXXXXXXXX@XXX.XXX.XXX.XXX:64773;rinstance=6502234ef65ea6e7 But, switching to the TCP i'm receiving the following warning: WARNING: registrar [sip_msg.c:194]: check_contacts(): contact uri is too long:[sip:XXXXXXXXX@192.168.0.1:60471;rinstance=bf4d7aae0f099bf0;transport=tcp] How can i solve this issue? Thank You.

1 1

How to send OPTIONS message by kamailio server
by 吴妙灵 27 Oct '17

27 Oct '17

Hello, I don't know how to send OPTIONS message from kamailio server to user client,can you give me same advice? Cheers, Ada

4 3

Timer function didnot trigger the routing function
by RAJA SREE RATHNA gomathinayagam 27 Oct '17

27 Oct '17

Hi, I tried using the timer function to call a function after specific time but it was not working as specified. Could you please help me with some example using timer to trigger a function. I tried it using the description given in below link. www.kamailio.org/docs/modules/devel/modules/timer.html Thanks and regards, Rathna.

2 3

Kamailio v5.0.4 Released
by Daniel-Constantin Mierla 27 Oct '17

27 Oct '17

Hello, Kamailio SIP Server v5.0.4 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.0.x versions are strongly recommended to be upgraded to v5.0.4. For more details about version 5.0.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/10/kamailio-v5-0-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 4

Set timeout before 180/183 reply arrived
by Yu Boot 27 Oct '17

27 Oct '17

Hello. Have a problem with some uplinks. They sometimes respond "100 Trying" on initial INVITE and... that's all. A caller waits for about 30 seconds and cancel a call deciding there's something wrong with our network. How to use failure route when 180/183 answers don't arrive in 5 or 10 seconds after INVITE?

2 2

Get value of a custom header param
by Arsen 26 Oct '17

26 Oct '17

Hi guys! Is it possible to get value of a custom parameter of the authorization header? I've checked selects in the textops mod and try the following: if (is_present_hf("Authorization")) { xplog("L_ERR", "custom auth $sel((a)hf_value.authorization.edatech_opaque) \n"); } Arsen Semionov www.eurolan.info cell: +442035198881

3 5

Upgrading from 1.5 to 3.x
by Brandon Shiers 26 Oct '17

26 Oct '17

I have a client that has an old Kamailio 1.5 installation. They would like to migrate and get off their old hardware and old OS. I've read the upgrade page for the 1.5.x migration to 3.0.0, are there any gotchas to be aware of other than what's there? Brandon Shiers, Network Engineer 937 West Main Street Riverton, WY 82501 307.857.6704 (o) 307.840.2366 (c) 307.856.1499 (f) brandon.shiers(a)cerento.com<mailto:Brandon.Shiers@cerento.com> [http://www.cerento.com/emailsignatures/logo.jpg]

2 1

Regarding rtimer usage under specific condition
by RAJA SREE RATHNA gomathinayagam 26 Oct '17

26 Oct '17

Hello, While going through the rtimer module we noticed that it execute the routing function repeatedly after the given interval of time. But our specification is to execute the routing function only once and timer should be enabled based on specific condition. Also there is no enable or disable option for rtimer module as in timer module, whether rtimer module can be enabled based on specific condition.If it is possible could you please suggest the solution. Thanks & Regards, Rathna.

2 1

shared dialogs
by : Paolo Visintin - Time-Net S.r.l. 26 Oct '17

26 Oct '17

Hi kamailio users, I'm wondering if there's something already done with shared dialog among more kamailio instances, in order to manage branches initalized by another kamailio instance. A ctually seems not possible (also with db mode) because kamailio is looking into caller/callee_sock and if it's different does not manage in a "standard" failover environment (master / slave) with vIP and keepalived and $fs = VIRTUAL_IP everything is working fine but in a distributed environment this could not happen as the relay is managed by a kamailio proxy Thanks in advance Paolo

4 7

I-CSCF Prefixes IMPU with + Symbol
by Ron McLeod 25 Oct '17

25 Oct '17

I am working with v5.0.3 and the IMS extensions. When I attempt to send a SIP MESSAGE from the application server towards the UE, the I-CSCF used to determine the S-CSCF for the subscriber by sending a Location-Info request to the HSS. I found that this request was failing because the IMPU (IMS Public User Identity) was being mangled by the I-CSCF, prefixing the user part with a + symbol. The example configuration file for the I-CSCF does in-fact does cause the r-uri to be prefixed with a + symbol if one is not already present. I'm wondering why since it breaks the LIR/LIA lookup. AS I-CSCF HSS | MESSAGE | | | ---------------------> | | | sip:ron@provider1.test | | | | | | | | | | LIR | | | ----------------------> | | | sip:+ron@provider1.test | | | | | | | | | DIAMETER_UNKNOWN_USER | | | <---------------------- | | | | | | | | 403 | | | <--------------------- | | | FORBIDDEN | | route{ #!ifdef WITH_DEBUG xlog("I-CSCF >>>>>>>>>>>>>>>>>>>> $rm $ru ($fu => $tu ($si:$sp) to $tu, $ci)\n"); #!endif if !($rU =~ "\+.*") { prefix("+"); } ... Thanks, Ron

1 0

Testing for user=phone with siputils uri_param() doesn't work
by George Diamantopoulos 25 Oct '17

25 Oct '17

Hello, In my script I've had to test for the user=phone parameter in URIs. I thought uri_param() from siputils would be handy for this operation on the RURI, so I did: if ( !uri_param("user","phone") ) { add_uri_param("user=phone"); } Unfortunately this wouldn't work (uri_param returns -1), and I end up with double user=phone params in the RURI. I was able to work around this by changing the test to: if ( !($(ru{uri.uparam}) == "phone") ) { add_uri_param("user=phone"); } But I wanted to ask if this is intentional or if I should file a bug or something. Thanks. George

3 3

Planning Kamailio v5.0.4
by Daniel-Constantin Mierla 25 Oct '17

25 Oct '17

Hello, I am considering to release Kamailio v5.0.4 later this week, likely on Wednesday or Tuesday. If you are aware of any issue in branch 5.0 which is not yet reported on github.com/kamailio/kamailio issues tracker, then do it as soon as possible to have a chance to look at it. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 5

usrloc module does not work with db_cluster
by Uri Shacked 25 Oct '17

25 Oct '17

Hi, I started using db_cluster module. DB are master-slave replication both ways. #!define DBURL "cluster://kamailio" when i stop the primary mysql server, all subscribers fails to register again. seems like the usrloc module does not work with the cluster. modparam("usrloc", "db_mode", 1) modparam("db_cluster", "cluster", "kamailio=>primary=9s9s;secondary=8s8s") it works OK for modules like dispatcher and htable to reload from DB. it shows the sql driver errors, but reloads from the secondary server. I need auto expire, so, p_usrloc is not a good option. working with version 4.2 any ideas ? Uri

2 1

FW: Mutual TLS with Skype for Business 2015
by Francisco Valentin Vinagrero 25 Oct '17

25 Oct '17

Hello, I'm trying to replace two old Audiocodes gateways (used to interconnect our Skype for Business infrastructure to the PSTN) with a new Kamailio cluster. I am having some trouble to get the TLS mutual authentication working with Kamailio. For the moment, I'm just trying to receive the incoming OPTIONS from SfB, but I get all the time certificate verification errors: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR: <core> [tcp_read.c:1330]: tcp_read_req(): ERROR: tcp_read_req: error reading My tls.cfg is quite simple, with the same config for client and server (and one single listen=tls:<my IP>:5061 in the Kamailio.cfg file) [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem # => This certificate's Subject is the DNS alias for the cluster, with all the kamailios in the cluster as Subject Alternative Names ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem # => Kamailio and Skype for Business are signed by different CAs, so here I concatenated all intermediate and root CAs [client:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/tls/key_gw_sfb.pem certificate = /usr/local/etc/kamailio/tls/cert_gw_sfb.pem ca_list = /usr/local/etc/kamailio/tls/myca_and_sfbca.pem When I run Kamailio, I can see incoming OPTIONS from Microsoft Exchange Unified Messaging (UM), whose certificate I verify without any issues. UM presents a certificate issued for a single machine, so no Subject Alternative Names (SANs) are involved. The problem comes with the TLS handshake for the Skype Mediation pool. They have a certificate with Subject = DNS alias and all the physical machines that are behind the alias appear listed as Subject Alternative Names (SANs) in the certificate. As the only difference between UM and Skype's Mediation is the certificate's Subject, I think I am missing something on my configuration to validate the SANs instead of the subject. Is the TLS module doing any reverse DNS lookup to verify this? Thanks, Francisco.

1 0

Advice on taking the best path to build a Presence & IM server
by José Seabra 24 Oct '17

24 Oct '17

Hello there, I'm writing here in order to get your experiences about a Presence & IM server. I would like to know what is the best solution to follow: - SIP Simple or XMPP? I've more knowledge about SIP Simple with kamailio but i did some researches and i found several open sources unified communications software that's implement Presence and IM using an XMPP server. So what is your opinion about which is the best solution to choose, taking in considerations the following aspects: 1. scalability 2. interopeability 3. permornace 4. features(like: Personal assistance, voicemail integration, chat group, integration with social media) Should i choose Kamailio SIP Simple with xcap or an XMPP server(Prosody or ejabberd, etc) I will appreciate a lot your opinion. Thank you Regards José -- Cumprimentos José Seabra

3 4

lcr
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

(no subject)
by Abdoul Osséni 24 Oct '17

24 Oct '17

Hello, Is there a LCR module based on ACD or QoS (ie. LCR based on terminated calls (not setups)? Thank you in advance. Best regards Abdoul OSSENI

1 0

How to check 4XX/5XX response in general.
by vivek＠advaitamtech.com 24 Oct '17

24 Oct '17

Hi, Is there any way to catch all 4XX and 5XX SIP error codes rather then specifying individually ie. 404, 500, 400 etc in kamailio. Thanks, Vivek.

2 2

NatHelper ignoring force_socket module parameter
by Asgaroth 24 Oct '17

24 Oct '17

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 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, F_MALLOC, DBG_F_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: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

2 5

No music on hold with mohqueue
by Fedor 24 Oct '17

24 Oct '17

I got in my kamailio.cfg: loadmodule "mohqueue.so" # --- music on hold moh ---- modparam ("mohqueue", "db_url", "mysql://kamailio:PASSWORDHERE@localhost/ kamailio") modparam ("mohqueue", "mohdir", "/etc/kamailio/moh") modparam ("mohqueue", "moh_maxcalls", 3000) Databases for mohqueue exist rttproxy configured fed@vps1:/etc/kamailio# ll /etc/kamailio/moh/ total 185424 drwxr-xr-x 2 kamailio kamailio 4096 Oct 8 09:32 ./ drwxr-xr-x 3 root root 4096 Oct 8 10:04 ../ -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.0 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.1 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.10 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.11 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.12 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.13 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.14 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.15 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.16 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.17 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.18 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.19 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.2 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.20 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.3 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.4 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.5 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.6 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.7 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.8 -rw-r--r-- 1 kamailio kamailio 8627660 Oct 8 09:32 hold.9 -rw-r--r-- 1 kamailio kamailio 8627660 Jan 2 2017 hold.wav rtpproxy configured, but no music on hold(srtp works fine) Can anybody help me, please?

2 1

Multi crashes in the same day
by igor.potjevlesch＠gmail.com 23 Oct '17

23 Oct '17

Hello ! I have multi crashes on a Kamailio 4.2.8 instance. Already 6 coredump today, same time with the same scenario. Below the output of the last coredump: (gdb) where #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 #2 0x000000000061aad2 in qm_debug_frag (qm=0x7f9a8a7aa010, f=0x7f9a8aca5968) at mem/q_malloc.c:161 #3 0x000000000061bf1c in qm_malloc (qm=0x7f9a8a7aa010, size=24, file=0x7131d0 "<core>: re.c", func=0x71483e "subst_run", line=440) at mem/q_malloc.c:388 #4 0x0000000000505e2c in subst_run (se=0x7f9a8ac08b78, input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, count=0x0) at re.c:440 #5 0x0000000000506fff in subst_str (input=0x7f9a8aca5918 "sip:0123456789@customer.local", msg=0x7f9a8abfef38, se=0x7f9a8ac08b78, count=0x0) at re.c:515 #6 0x00007f9a87ff629d in subst_uri_f (msg=0x7f9a8abfef38, subst=0x7f9a8ac08b78 "X\215\300\212\232\177", ignored=0x0) at textops.c:744 #7 0x000000000041d51b in do_action (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1094 #8 0x0000000000429ba3 in run_actions (h=0x7fffed0800d0, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1583 #9 0x000000000042a208 in run_actions_safe (h=0x7fffed081380, a=0x7f9a8ab64cc8, msg=0x7f9a8abfef38) at action.c:1648 #10 0x0000000000542994 in rval_get_int (h=0x7fffed081380, msg=0x7f9a8abfef38, i=0x7fffed080890, rv=0x7f9a8ab64ef8, cache=0x0) at rvalue.c:924 #11 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed081380, msg=0x7f9a8abfef38, res=0x7fffed080890, rve=0x7f9a8ab64ef0) at rvalue.c:1918 #12 0x000000000041cf77 in do_action (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1064 #13 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab66a30, msg=0x7f9a8abfef38) at action.c:1583 #14 0x000000000041d3f6 in do_action (h=0x7fffed081380, a=0x7f9a8ab6b070, msg=0x7f9a8abfef38) at action.c:1079 #15 0x0000000000429ba3 in run_actions (h=0x7fffed081380, a=0x7f9a8ab607b8, msg=0x7f9a8abfef38) at action.c:1583 #16 0x000000000042a2d0 in run_top_route (a=0x7f9a8ab607b8, msg=0x7f9a8abfef38, c=0x7fffed081380) at action.c:1669 #17 0x00007f9a891f6d0f in prepare_new_uac (t=0x7f9a6de86518, i_req=0x7f9a8abfef38, branch=0, uri=0x7fffed0814d0, path=0x7fffed0814b0, next_hop=0x7f9a8abff1a8, fsocket=0x0, snd_flags=..., fproto=0, flags=2, instance=0x7fffed0814a0, ruid=0x7fffed081490, location_ua=0x7fffed081480) at t_fwd.c:410 #18 0x00007f9a891faded in add_uac (t=0x7f9a6de86518, request=0x7f9a8abfef38, uri=0x7f9a8abff1a8, next_hop=0x7f9a8abff1a8, path=0x7f9a8abff568, proxy=0x0, fsocket=0x0, snd_flags=..., proto=0, flags=2, instance=0x7f9a8abff578, ruid=0x7f9a8abff590, location_ua=0x7f9a8abff5a0) at t_fwd.c:855 #19 0x00007f9a89201ddd in t_forward_nonack (t=0x7f9a6de86518, p_msg=0x7f9a8abfef38, proxy=0x0, proto=0) at t_fwd.c:1723 #20 0x00007f9a891ee374 in t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, proto=0, replicate=0) at t_funcs.c:355 #21 0x00007f9a8922bd89 in _w_t_relay_to (p_msg=0x7f9a8abfef38, proxy=0x0, force_proto=0) at tm.c:1517 #22 0x00007f9a8922ceee in w_t_relay (p_msg=0x7f9a8abfef38, _foo=0x0, _bar=0x0) at tm.c:1718 #23 0x000000000041d48d in do_action (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1088 #24 0x0000000000429ba3 in run_actions (h=0x7fffed082190, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1583 #25 0x000000000042a208 in run_actions_safe (h=0x7fffed0848f0, a=0x7f9a8a83e878, msg=0x7f9a8abfef38) at action.c:1648 #26 0x0000000000542994 in rval_get_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, i=0x7fffed082668, rv=0x7f9a8a83eb60, cache=0x0) at rvalue.c:924 #27 0x0000000000546bcc in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082668, rve=0x7f9a8a83eb58) at rvalue.c:1918 #28 0x0000000000546fc2 in rval_expr_eval_int (h=0x7fffed0848f0, msg=0x7f9a8abfef38, res=0x7fffed082af0, rve=0x7f9a8a83f280) at rvalue.c:1926 #29 0x000000000041cf77 in do_action (h=0x7fffed0848f0, a=0x7f9a8a83fbd0, msg=0x7f9a8abfef38) at action.c:1064 #30 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a838c88, msg=0x7f9a8abfef38) at action.c:1583 #31 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996598, msg=0x7f9a8abfef38) at action.c:712 #32 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a9955b8, msg=0x7f9a8abfef38) at action.c:1583 #33 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a996a50, msg=0x7f9a8abfef38) at action.c:1079 #34 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a8fa068, msg=0x7f9a8abfef38) at action.c:1583 #35 0x0000000000419f13 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:712 #36 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a830c70, msg=0x7f9a8abfef38) at action.c:1583 #37 0x000000000041d3f6 in do_action (h=0x7fffed0848f0, a=0x7f9a8a830db8, msg=0x7f9a8abfef38) at action.c:1079 #38 0x0000000000429ba3 in run_actions (h=0x7fffed0848f0, a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38) at action.c:1583 #39 0x000000000042a2d0 in run_top_route (a=0x7f9a8a81e7d0, msg=0x7f9a8abfef38, c=0x0) at action.c:1669 #40 0x00000000005095b6 in receive_msg ( buf=0xa78b60 "INVITE sip:0123456789@A.B.C.181:5060;transport=UDP;user=phone SIP/2.0\r\nFrom: <sip:0987654321@A.B.C.148:5060>;tag=BN1507883839-1-1507047459-332794206;tran sport=UDP\r\nTo: <sip:0123456789@H.G.F.D"..., len=1119, rcv_info=0x7fffed084be0) at receive.c:216 #41 0x000000000060945a in udp_rcv_loop () at udp_server.c:521 #42 0x00000000004a6e62 in main_loop () at main.c:1629 #43 0x00000000004acfa6 in main (argc=7, argv=0x7fffed085018) at main.c:2581 Regards, Igor.

2 2

Presence WARNING after publish and no NOTIFY
by Yuriy Gorlichenko 23 Oct '17

23 Oct '17

Hi use 5.0.1 with presence for TCP/UDP endpoints presence works PERFECT but for WS socket presence i see next (small cleaned log here with all described bellow https://pastebin.com/8a77ggj0 ): When WS endpoint subscribes - kamailio After kamailio sends 202 to SUBSCRIBE message (see it on client) it sends fast NOTIFY correcly without any trouble, and i see at the kamailio reply from WS endpoint - 200 WS endpoint added to active_watchers table. But when Subscribed endpoint makes for example some call and presence server sends PUBLISH i see: presence [presentity.c:426]: delete_presentity_if_dialog_id_exists(): Presentity already exists - deleting it At the logs and then presence server doesn't send NOTIFY to WS endpoint What is the issue of this? Can not understand why it happens. Can someone to help me understand this? And the dump i see that kamailio send first NOTIFY form UDP port but then translates it to tls for websocket and i see correct NOTIFY at the WS endpoint

2 4

MoS(QoS) measurement in Kamailio based IMS network
by Kranti Kumar 23 Oct '17

23 Oct '17

Hi All, I am using Kamailio IMS VMimage which is provided by Franz. I am successfully established the call between the IMS clients. Here I would like to measure the MoS mean opnion score (QoS) of Calls (Voice quality). Can any one suggest me / guide me how to measure the same in this IMS network. Thank you so much in advance. Regards, -kranti

2 1

how to bridge two landline users
by 赵国杰 23 Oct '17

23 Oct '17

Hello, In a standard sip flow, the call goes like: sip user A --> kamailio --> pstn --> landline user B. However, when user A has a bad internet access, the audio is broken. So what I want is to let sip user A send a invite to kamailio first, then kamailio send invite to user A and B's landline number through pstn, then bridge the two call together. I understand this can be achieved by using FREESWITCH originate and bridge command. I've tried but there's no audio both ways, which really makes me feel stupid of myself. So I'm wondering if this can be done with kamailio? If so, how? Thanks Jesse

2 1

uac_replace_from and header adulterations
by Alex Balashov 23 Oct '17

23 Oct '17

Hello, I've got a bizarre problem caused by bad UA behaviour: UA A ---> Kamailio (P) ---> UA B 1. UA A sends initial INVITE through P to B; 2. Kamailio (P) makes some modifications to the From header using uac_replace_from() and passes along to B. 3. B sends an in-dialog request (e.g. BYE or reinvite) to A through P; in doing so, it modifies the To (formerly From) value slightly, replacing the hostname portion in the To URI with a different value to the one that was received in the From header. 4. Kamailio relays this in-dialog request to A, but with an adulterated/clipped/truncated/grammatically invalid To header now. 5. A responds with 400 Bad Request due to invalid To header. I would not dispute that UA B should not be modifying the remote URI in this manner. But since it does, it gives rise to two questions: 1. Why does Kamailio not simply discard the modified To header and restore the original value stored in the Record-Route rider parameter? Is it because the Record-Route parameter does not contain the original header value, but rather some data complementary to the current header value? Or is it because the UAC code takes a checksum of the original remote URI header value and stores it, and just checks it when restoring on principle? If so, what's the motive for that? 2. Wouldn't it be better behaviour to simply reject a request so malformed, rather than passing it on with a corrupt restored value? If Kamailio can detect that the header has been tampered with, why pass it on? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Help on Video Conference
by Kranti Kumar 23 Oct '17

23 Oct '17

Hello , I’m trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ My configuration details: HSS - 10.0.0.9 (./hss.sh) PCSCF - 10.0.0.10 (kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg) ICSCF - 10.0.0.11(kamailio -f /usr/local/etc/kamailio/kamailio-icscf.cfg) SCSCF - 10.0.0.12 (kamailio -f /usr/local/etc/kamailio/kamailio-scscf.cfg) host machine: 10.0.0.5 I am able to register mobile clients ( IMSdroid) and make voice/video calls between them successfully. Now I would like to make a conference call among 3 IMSdroid users but its failing. Could you please suggest me the way forward here. I am thinking to integrate Application server (AS) to enable Video Conferences. Could you please guide me how to enable AS in this VM image and configuration details. Is there any possibility to do Video conference using this current setup(VM image). Could you please suggest me. Thank you so much in advance. Regards, -kranti

2 1

Kamailio IMS deployment
by Ramya Y 22 Oct '17

22 Oct '17

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

5 22

call_id in dialog module
by ‪mahsa _a_‬ ‪ 20 Oct '17

20 Oct '17

hi all;i want to write "billing" project with kamailio,i using python,postgresql, dialog module and falcon.my problem is in "call_id" section.my code(this section) is in python : """... if (method == 'INVITE'): callid = req.get_param_as_list("call_id") print(req.get_param_as_list("Ts")) print("call id = %s " % callid) elif (method == 'ACK'): time_req = int(req.get_param_as_list("Ts")[0]) callid = req.get_param_as_list("call_id")[0] print("user %s call to %s at time %d % (from_user,to_user,time_req))... """ and in kamailio.cfg: """http_query("http://127.0.0.1:5000/test?method=$rm& call_id=$dlg(callid)&Ts=$Ts&fu=$fu&tu=$tu","$var(result)"); """" "callid" in ACK is ok but "callid' in invite is "null", where is my problem? thanks.

2 1

Retransmission handling
by Volkan Oransoy 19 Oct '17

19 Oct '17

Hi all, I wonder purpose of this block in default script. What I didn’t understand is the purpose of "t_check_trans()”. Accoording to documentation, this method returns a boolean. So why do we use it even we don’t check output? # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } /Volkan

3 3

VoLTE Capable LTE Handsets which support IMS uisng IPv4
by Ron 19 Oct '17

19 Oct '17

I would like to support VoLTE using an IPv4 IMS network rather than IPv6 network. The handsets that I have been trying insist on the IMS network support either IPv6 or IPv4v6 dual stack. After giving them IPv4v6 connectivity, I am finding that even though the handsets asked for both IPv4 and IPv6 P-CSCFs, they only try the IPv6 proxy (if only the IPv4 proxy is provided, they will not try at all). I am wondering of anyone has worked with any LTE handsets which are willing to support IMS using IPv4 only. Thanks, Ron

3 2

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