- sr-users - kamailio.org

Issue with linking contact to IMPU
by Rafi Aizenstain 01 Jan '18

01 Jan '18

Hi there, I need some help here to understand why the query in [usrloc_db.c:1084] is empty. This causes the linking of contact to IMPU to fail. I'm using Kamailio 5.0.4. 10(13882) DEBUG: ims_usrloc_scscf [impurecord.c:1284]: unref_subscription_unsafe(): un-reffing subscription [rafi(a)mnc001.mcc001.3gppnetwork.org] - was [3] 10(13882) DEBUG: ims_registrar_scscf [save.c:519]: update_contacts_helper(): updating the contacts for IMPU <sip:rafi@mnc001.mcc001.3gppnetwork.org> 10(13882) DEBUG: ims_registrar_scscf [save.c:137]: calc_contact_expires(): Calculated expires for contact is 600000 10(13882) DEBUG: ims_registrar_scscf [save.c:548]: update_contacts_helper(): Need to update contact: <sip:rafi@192.168.10.48:57230;transport=udp>: q_value [-1],sos: [0],expires [600000] 10(13882) DEBUG: ims_registrar_scscf [save.c:550]: update_contacts_helper(): packing contact information 10(13882) DEBUG: ims_registrar_scscf [save.c:556]: update_contacts_helper(): adding/updating contact based on prior existence 10(13882) DEBUG: ims_usrloc_scscf [impurecord.c:748]: get_scontact(): looking for contact [sip:rafi@192.168.10.48:57230;transport=udp] in slot 300 10(13882) DEBUG: ims_registrar_scscf [save.c:562]: update_contacts_helper(): inserting new contact 10(13882) DEBUG: ims_usrloc_scscf [impurecord.c:562]: insert_scontact(): INSERTing ucontact in usrloc module 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [+g.3gpp.cs-voice] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [+g.3gpp.icsi-ref] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [audio] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [+g.oma.sip-im.large-message] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [+g.3gpp.smsip] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [language] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [+g.oma.sip-im] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:118]: new_ucontact(): Checking param [expires] 10(13882) DEBUG: ims_usrloc_scscf [ucontact.c:173]: new_ucontact(): generating hash based on [sip:rafi@192.168.10.48:57230;transport=udp] 10(13882) DEBUG: ims_usrloc_scscf [impurecord.c:280]: mem_insert_scontact(): Created new contact in memory with AOR: [sip:rafi@192.168.10.48:57230;transport=udp] and hash [300] 10(13882) DEBUG: ims_usrloc_scscf [usrloc_db.c:289]: db_insert_ucontact(): DB: inserting ucontact [sip:rafi@192.168.10.48:57230;transport=udp] 10(13882) DEBUG: ims_usrloc_scscf [usrloc_db.c:311]: db_insert_ucontact(): Converted params to string to insert into db: [+g.3gpp.cs-voice;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs;audio;+g.oma.sip-im.large-message;+g.3gpp.smsip;language=en,fr;+g.oma.sip-im;expires=600000] 10(13882) DEBUG: ims_usrloc_scscf [udomain.c:843]: ref_contact_unsafe(): incrementing ref count on contact [sip:rafi@192.168.10.48:57230;transport=udp], was 1 10(13882) DEBUG: ims_usrloc_scscf [impurecord.c:1225]: link_contact_to_impu(): number of contacts for IMPU [sip:rafi@mnc001.mcc001.3gppnetwork.org] is 1 10(13882) DEBUG: ims_usrloc_scscf [usrloc_db.c:1065]: db_link_contact_to_impu(): DB: linking contact to IMPU 10(13882) DEBUG: ims_usrloc_scscf [usrloc_db.c:1084]: db_link_contact_to_impu(): QUERY IS [] and len is 0 10(13882) ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Query was empty (1065) 10(13882) ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query 10(13882) ERROR: ims_usrloc_scscf [usrloc_db.c:1086]: db_link_contact_to_impu(): Unable to link impu-contact in DB - impu [sip:rafi@mnc001.mcc001.3gppnetwork.org], contact [sip:rafi@192.168.10.48:57230;transport=udp] 10(13882) ERROR: ims_usrloc_scscf [impurecord.c:1227]: link_contact_to_impu(): Failed to update DB linking contact [sip:rafi@192.168.10.48:57230;transport=udp] to IMPU [sip:rafi@mnc001.mcc001.3gppnetwork.org]...continuing but db will be out of sync! 10(13882) DEBUG: ims_usrloc_scscf [udomain.c:852]: unref_contact_unsafe(): decrementing ref count on contact [sip:rafi@192.168.10.48:57230;transport=udp], was 2 Best Regards, Rafael A.

1 0

error running icscf module on 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

2 1

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

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users