- sr-users - kamailio.org

Breaking HTABLE changes from v5.2 to v5.3? Function renamed
by Joel Serrano 08 Aug '19

08 Aug '19

Hi all, Just noticed that when updating from 5.2 to 5.3 I saw some errors in Kam logfile regarding HTABLE module commands failing, the reason is the function 'sht_has_name' has been renamed to 'sht_match_name': https://www.kamailio.org/docs/modules/stable/modules/htable.html#htable.f.s… https://www.kamailio.org/docs/modules/devel/modules/htable.html#htable.f.sh… Not sure if updating this doc is automatic or manual but I think it's definitely worth mentioning: https://www.kamailio.org/wiki/install/upgrade/stable-to-devel Just FYI! Cheers, Joel.

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Re: [SR-Users] Sqlops
by arish haque 08 Aug '19

08 Aug '19

Hi, Passing the value within single quotes works. Thanks for the help Fred. With Regards, Arish

1 0

Reload geoip2 database without restart
by Joel Serrano 07 Aug '19

07 Aug '19

Hi guys, Is there a way to instruct the geoip2 module to reload its database without a full restart of kamailio? Maybe some rpc command or something? I haven't found anything in the docs other than the comment about "Database is loaded at startup in cache." which makes me assume that just overwriting the db file wouldn't be enough. Any suggestions? Thanks!! Joel.

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Sipt redirection number retrieves old values
by Duarte Rocha 07 Aug '19

07 Aug '19

Greetings, I'm doing VoiceMail detection using $sipt(redirection_number) from SIPT module and i've been having some false positives on the detection. I inspected further and noticed that in some cases $sipt(redirection_number) returns old values from previous messages, which makes me think that it's value is not being correctly refreshed. I'm using this variable on a on_reply_route. As far as i know, $sipt(redirection_number) is read only and can't be refreshed by me in the code. Here is an example : Aug 7 16:51:49 dev_sbc_a_vts01 /usr/local/kamailio-5.2/sbin/kamailio[4746]: ERROR: <script>: VMD2 : Reply Code : 180 ; Sipt Redir = 9610050000 and Call ID = 00158FED-98B7-E911-8B3B-4DA50AE2ED72(a)192.168.24.86 and Cseq = 83 Here is the JSON from the 180 message : https://pastebin.com/zFHKxyav As you can see, there isn't any 9610050000 in the message. What can be causing this issue ? Best Regards, Duarte Rocha

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Sqlops
by arish haque 07 Aug '19

07 Aug '19

Hi all, I am querying mysql db inside routing block in kamailio.cfg ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Unknown column '8d77' in 'where clause' (1054) Aug 7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Aug 7 18:09:22 localhost /usr/local/sbin/kamailio[25333]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [select * from vca_asterisk where front_id=8d77] I'm getting the above error message when, variable front_id=8d77 is alphanumeric. It's working fine when front_id is pure numeric. eg: front_id=877 Database column front_id is VARCHAR(255). Any suggestions to solve this error? Thanks, Arish

2 1

ul.rm_contact usage
by Володимир Іванець 07 Aug '19

07 Aug '19

Hello, I'm trying to use *ul.rm_contact* RPC command to unregister spesific Contact but can not figure out "contact" parameter. I tried different combinations but received "error: 404 - Contact not found" message every time. Could please someone tell how this command should be called for one of registered Contacts? Thank you! *# kamcmd ul.dump* *{* * Domains: {* * Domain: {* * Domain: location* * Size: 1024* * AoRs: {* * Info: {* * AoR: 200-test* * HashID: 406934013* * Contacts: {* * Contact: {* * Address: sip:200-test@1.1.1.1:52181;transport=TCP;rinstance=5d1a0b6b677f269d* * Expires: 114* * Q: -1.000000* * Call-ID: YmNlMmJlNDMwMGY5NGFkYzkzOTU2NDFhNzc5NWE3Mzc.* * CSeq: 4* * User-Agent: 3CXPhone 6.0.26523.0* * Received: sip:1.1.1.1:52181;transport=tcp* * Path: [not set]* * State: CS_NEW* * Flags: 0* * CFlags: 65536* * Socket: tcp:172.16.30.205:5060 <http://172.16.30.205:5060>* * Methods: 5119* * Ruid: uloc-5d4983fb-40b5-4* * Instance: [not set]* * Reg-Id: 0* * Server-Id: 0* * Tcpconn-Id: 3* * Keepalive: 0* * Last-Keepalive: 1565099252* * Last-Modified: 1565099252* * }* * Contact: {* * Address: sip:200-test@1.1.1.1:58166;rinstance=5996b96740d52a3a* * Expires: 118* * Q: -1.000000* * Call-ID: 97566MWMxYmVhMmE1OGUyOTlkMzVmYTYzZmE2ZDE5YWJkYTc* * CSeq: 4* * User-Agent: X-Lite release 5.5.0 stamp 97566* * Received: sip:1.1.1.1:58166 <http://1.1.1.1:58166>* * Path: [not set]* * State: CS_NEW* * Flags: 0* * CFlags: 65536* * Socket: udp:172.16.30.205:5060 <http://172.16.30.205:5060>* * Methods: 5087* * Ruid: uloc-5d4983fb-4093-2* * Instance: [not set]* * Reg-Id: 0* * Server-Id: 0* * Tcpconn-Id: -1* * Keepalive: 0* * Last-Keepalive: 1565099256* * Last-Modified: 1565099256* * }* * }* * }* * }* * Stats: {* * Records: 1* * Max-Slots: 1* * }* * }* * }* *}*

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100% CPU usage after NTP date update
by Benjamin Marty 07 Aug '19

07 Aug '19

Hello, I'm facing the same issue. Is there a solution for this in the meantime? Benjamin https://lists.kamailio.org/pipermail/sr-users/2015-November/090487.html

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Problem with tcp connections
by logosvlad＠gmail.com 07 Aug '19

07 Aug '19

Hello! we have kamailio 5.2.3 and openssl 1.1.0f. wss connections incoming to kamailio and outgoing sip connections, after some time http request check not getting answer from kamailio. Netstat show increase recieve tcp buffer on connections of kamailio. We set up script which after not response of http check, run SIGSEGV of main process of kamailio. gdb) bt full #0 0x00007fb779279450 in futex_wait (private=<optimized out>, expected=1, futex_word=0x7fb6f78cb964) at ../sysdeps/unix/sysv/linux/futex-internal.h:61 __ret = -512 err = <optimized out> #1 futex_wait_simple (private=<optimized out>, expected=1, futex_word=0x7fb6f78cb964) at ../sysdeps/nptl/futex-internal.h:135 No locals. #2 __pthread_rwlock_wrlock_slow (rwlock=0x7fb6f78cb958) at pthread_rwlock_wrlock.c:67 waitval = 1 result = 0 futex_shared = <optimized out> #3 0x00007fb77995eee9 in CRYPTO_THREAD_write_lock () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 No symbol table info available. #4 0x00007fb779907896 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 No symbol table info available. #5 0x00007fb77990807e in CRYPTO_free_ex_data () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 No symbol table info available. #6 0x00007fb779c65a09 in SSL_free () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 No symbol table info available. #7 0x00007fb777c022d6 in tls_h_tcpconn_clean (c=0x7fb6fbb73d58) at tls_server.c:655 extra = 0x7fb6fba1a6d0 __func__ = "tls_h_tcpconn_clean" #8 0x0000559ddff2966f in _tcpconn_free (c=0x7fb6fbb73d58) at core/tcp_main.c:1339 __func__ = "_tcpconn_free" #9 0x0000559ddff3c5f7 in tcpconn_put_destroy (tcpconn=0x7fb6fbb73d58) at core/tcp_main.c:3017 __func__ = "tcpconn_put_destroy" #10 0x0000559ddff3e0f2 in send_fd_queue_run (q=0x559de0518bc0 <send2child_q>) at core/tcp_main.c:3210 p = 0x7fb77e19e2c8 t = 0x7fb77e19e2c8 __func__ = "send_fd_queue_run" #11 0x0000559ddff4f61c in tcp_main_loop () at core/tcp_main.c:4548 si = 0x0 r = 10 __func__ = "tcp_main_loop" #12 0x0000559ddfe4e85d in main_loop () at main.c:1753 i = 3 pid = 0 si = 0x0 si_desc = "udp receiver child=2 sock=127.0.0.1:5060 (127.0.0.1:5060)\000\065\060\066\060)\000\000\000\000\200\000\000\000\000\000\000\000\002\000\000\000\000\006\000\000\000\000\000\000\000O \254\177\267\177\000\000p\324\067?\375\177\000\000\060\000\000\000\060\000\000\000\020\325\067?\375\177\000\000\360k\255}\267\177\000" nrprocs = 3 woneinit = 1 __func__ = "main_loop" #13 0x0000559ddfe554ce in main (argc=13, argv=0x7ffd3f37d7a8) at main.c:2696 cfg_stream = 0x559de1c10010 c = -1 r = 0 tmp = 0x7ffd3f37deba "" tmp_len = 32765 port = 1060623840 proto = 32695 options = 0x559de01c2a50 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 33232859 ---Type <return> to continue, or q <return> to quit--- rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0xf0b5ff <error: Cannot access memory at address 0xf0b5ff> st = {st_dev = 19, st_ino = 13222, st_nlink = 2, st_mode = 16832, st_uid = 108, st_gid = 112, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = { tv_sec = 1560780032, tv_nsec = 539855841}, st_mtim = {tv_sec = 1565076521, tv_nsec = 998570962}, st_ctim = {tv_sec = 1565076521, tv_nsec = 998570962}, __glibc_reserved = {0, 0, 0}} __func__ = "main" Maybe our problem correlate with https://github.com/kamailio/kamailio/issues/1860

1 0

app_ruby - invalid ruby environment attributes or parameters
by Andrew White 07 Aug '19

07 Aug '19

Hey all, I’m setting up a dev-7 (bd477f) server to test as per the other thread, however I’m getting the following loading my config: ERROR: app_ruby [app_ruby_api.c:465]: sr_kemi_ruby_exec_func_ex(): invalid ruby environment attributes or parameters I’m unsure what this error means - are there expected environment variables or other values that must be set? I didn’t see this issue in dev-4. I thought it might be not able to find the library, but I ran with debug=4 and saw: DEBUG: app_ruby [app_ruby_api.c:193]: ruby_sr_init_child(): test execution without error ERROR: app_ruby [app_ruby_api.c:107]: app_ruby_print_last_exception(): exception: undefined method `>' for false:FalseClass I changed kamailio.cfg to load my generic routing script, and during the onreply_manage it runs: scode = KSR::PV.get("$rs”) This now returns false (which appears to be the second error above, I check if scode is > 599). Has anything changed in the way ruby is loaded in dev-7? I’m using the same ruby/ruby-devel library versions on the same OS. I did this deploy the same with devel-4 a few weeks ago and this didn’t happen. Thanks! Andrew

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using in almost one worker
by logosvlad＠gmail.com 06 Aug '19

06 Aug '19

Hello! We have in configuration : c <sr-users(a)lists.kamailio.org>hildren=3 tcp_children=30 But in 90-95% connections kamailio use one worker (one process) by statitistics netstat -anp. Why not using more other workers (processes) ?

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what version openssl recomnded for kamailio 5.2.3
by logosvlad＠gmail.com 06 Aug '19

06 Aug '19

Hello! what version openssl 1.1.x did you recomnded for kamailio 5.2.3 for more stable operation ?

1 0

The domain module in 5.1.8 has a memory leak
by Mack Hendricks 06 Aug '19

06 Aug '19

Based on my tests, the memory leak is only exposed when you are using domain attributes, which are stored in the domain_attrs table. I wanted to see if anymore else is running into this issue. *Reproducible use case:* 1) Insert the following into the database insert into domain values (null,"aprilco.com","aprilco.com",now()); insert into domain_attrs values (null,"aprilco.com","pbx_type",0,"0",now()); insert into domain_attrs values (null,"aprilco.com","pbx_id",0,"65",now()); 2) Open a new terminal window and execute: watch "kamcmd core.shmmem" 3) Open up another terminal windows and execute: while true; do kamcmd domain.reload; done *Expected Results:* You will see the shared memory being allocated, but never freed. The more entries you have in the domain_attrs table, the faster it will run out of memory. Thanks, *Mack Hendricks / Head of Support / dOpenSource* web: http://dopensource.com support: +888-907-2085 dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services

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Error Registering the user and finding the AVP
by jespinosa 06 Aug '19

06 Aug '19

Hello, My name is Javier, I must first apologize for my little knowledge about the tool and my poor English. I am trying to connect a functional core network LTE with Kamailio and I have used the Polaris Networks Solution that brings an iso with both functionalities. The EPC is correctly configured and working because I can have data connection in my UE, I have also activated all the IMS options in the EPC: I have created the APN with ims identifier, the default carrier QCI assigned is 5 (for SIP signaling), the MME has activated the option to support IMS Voice over the PS session and the HSS and PCRF nodes are configured with the interfaces at the addresses and ports Cx-HSS 127.0.0.1:3868 and Cx-PCRF 10.11.0.35:3868 . There are several parameters that I do not know if it is necessary to activate, for example, in the PCRF the customization of the QCI for the media type, if I must include the OCS and OFCS as nodes, or if in the profile of my APN I must include authentication by PDN AAA. The configuration I use for the IMS is as follows: imsrealm = net1.test imsport = 5060 pcscfgmip = 10.77.0.35 pcscfgmipintf = eth6 pcscfrxip = 10.102.81.111 pcscfrxipintf = eth1 icscfmwip = 127.0.0.21 icscfmwipintf = lo:21 icscfcxip = 127.0.0.20 icscfcxipintf = lo:20 scscfmwip = 127.0.0.23 scscfmwipintf = lo:23 scscfcxip = 127.0.0.22 scscfcxipintf = lo:22 audiobandwidth = 64 videobandwidth = 128 epcrealm = epc.mnc001.mcc001.3gppnetwork.org pcrfhost = pcrf2.morsecore pcrfrxip = 10.11.0.35 pcrfrxport = 3868 hsshost = hss6.morsecore hsscxdxip = 127.0.0.1 hsscxdxport = 3868 Both cores are running on the same machine but I have disabled the option of PCRF Co-located with IMS to be able to edit the pcrf host name since if I enable it it does not reach the peer pcrf.localhost even if the DNS is well configured. My UE is not able to launch the REGISTER by itself when it is going to make a call, so I am using the Google IMSDroid application to be able to launch the SIP requests and check the operation of the IMS. In the application of the UE the configuration is as follows: Identity: Alice Public Identity: sip:<MSISDN>@net1.test (MSISDN=8800000451 configurado previamente en el EPC) Private Identity: sip:<IMSI>@net1.test (IMSI incluyendo MCC y MNC) Password: alice Realm: net1.test (Aunque no sé si debo poner aquí epc.mnc001.mcc001.3gppnetwork.org, pero entonces la petición da error) Network: Proxy-CSCF Host: 10.77.0.35:5060 Transport UDP and NO Proxy-CSCF discovery. So, I launch my request and the log of the P-CSCF node is as follows: Aug 2 12:50:04 morsecore kamailio[53656]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:04 morsecore kamailio[53656]: INFO: cdp [authstatemachine.c:200]: auth_client_statefull_sm_process(): after callback of event 1 Aug 2 12:50:04 morsecore kamailio[53802]: INFO: cdp [authstatemachine.c:200]: auth_client_statefull_sm_process(): after callback of event 7 Aug 2 12:50:04 morsecore kamailio[53655]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:05 morsecore kamailio[53664]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:07 morsecore kamailio[53659]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) Aug 2 12:50:11 morsecore kamailio[53666]: INFO: <script>: REGISTER (sip:8800000451@net1.test (10.77.0.196:37651) to sip:8800000451@net1.test, 76203fe6-1d8f-bcba-c9ab-4050ac0d8cc8) I understand that everything is correct until now, but in the I-CSCF node it appears: Aug 2 12:50:04 morsecore kamailio[53397]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_capabilities: Failed finding avp Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp Aug 2 12:50:04 morsecore kamailio[53465]: ERROR: ims_icscf [cxdx_uar.c:152]: async_cdp_uar_callback(): RC_IMS_DIAMETER_ERROR_USER_UNKNOWN Aug 2 12:50:04 morsecore kamailio[53465]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Aug 2 12:50:04 morsecore kamailio[53465]: ERROR: <script>: UAR failure - error response sent from module Where I believe that the REGISTER message is not even forwarded, however it seems that it gives an error that the user is unknown or not registered but if I access the MySQL database, the registered users are: +----+------------+-----------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+------------+-----------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | alice | net1.test | alice | | 91cc8c652ef5f37e91c468c7319d0086 | 044776963e2661788e8db076b393333e | NULL | | 2 | bob | net1.test | bob | | 6ef7f9c924f1385d83cf22cbb94cfbbb | 0d893d2d8c10a044b918e4bd73b6bee5 | NULL | | 3 | 8800000451 | net1.test | alice | | 46811c4e58f298953a6f9efccf6c32f1 | 49c7a7ad84fe797fbefa2185be4cd226 | NULL | +----+------------+-----------+----------+---------------+----------------------------------+----------------------------------+------+ However, I do not know if it is accessing the mounted MySQL database or accessing the HSS and on this node, in the databases it does not contain the user (although it must be because it is registered in the network correctly). In case it works, this is the trace of wireshark filtered by the protocols S1AP, SIP and Diameter just when a REGISTER request is launched: "596","2019-08-02 12:50:00.705985000","127.0.0.20","127.0.0.1","DIAMETER","130","cmd=Device-Watchdog Request(280) flags=R--- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "597","2019-08-02 12:50:00.706133000","127.0.0.1","127.0.0.20","DIAMETER","210","cmd=Device-Watchdog Answer(280) flags=---- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "599","2019-08-02 12:50:00.786878000","127.0.0.22","127.0.0.1","DIAMETER","130","cmd=Device-Watchdog Request(280) flags=R--- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "600","2019-08-02 12:50:00.787019000","127.0.0.1","127.0.0.22","DIAMETER","210","cmd=Device-Watchdog Answer(280) flags=---- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "602","2019-08-02 12:50:00.914405000","10.102.81.111","10.11.0.35","DIAMETER","130","cmd=Device-Watchdog Request(280) flags=R--- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "603","2019-08-02 12:50:00.914549000","10.11.0.35","10.102.81.111","DIAMETER","214","cmd=Device-Watchdog Answer(280) flags=---- appl=Diameter Common Messages(0) h2h=fedcb89 e2e=4df0bbf5 | " "622","2019-08-02 12:50:04.027238000","10.77.0.196","10.77.0.35","GTP <SIP>","1034","Request: REGISTER sip:net1.test (1 binding) | " "623","2019-08-02 12:50:04.027347000","10.77.0.196","10.77.0.35","GTP <SIP>","1034","Request: REGISTER sip:net1.test (1 binding) | " "624","2019-08-02 12:50:04.027448000","10.77.0.196","10.77.0.35","SIP","998","Request: REGISTER sip:net1.test (1 binding) | " "625","2019-08-02 12:50:04.028850000","10.102.81.111","10.11.0.35","DIAMETER","426","cmd=AA Request(265) flags=RP-- appl=3GPP Rx(16777236) h2h=fedcb8a e2e=4df0bbf6 | " "626","2019-08-02 12:50:04.029862000","10.11.0.35","10.102.81.111","DIAMETER","350","cmd=AA Answer(265) flags=-P-- appl=3GPP Rx(16777236) h2h=fedcb8a e2e=4df0bbf6 | " "630","2019-08-02 12:50:04.031113000","10.77.0.35","127.0.0.21","SIP","1280","Request: REGISTER sip:net1.test (1 binding) | " "631","2019-08-02 12:50:04.031794000","127.0.0.21","10.77.0.35","SIP","440","Status: 100 Trying | " "632","2019-08-02 12:50:04.032382000","127.0.0.20","127.0.0.1","DIAMETER","354","cmd=User-Authorization Request(300) flags=RP-- appl=3GPP Cx(16777216) h2h=fedcb8a e2e=4df0bbf6 | " "633","2019-08-02 12:50:04.033101000","127.0.0.1","127.0.0.20","DIAMETER","298","cmd=User-Authorization Answer(300) flags=-P-- appl=3GPP Cx(16777216) h2h=fedcb8a e2e=4df0bbf6 | " "636","2019-08-02 12:50:04.033557000","127.0.0.21","10.77.0.35","SIP","504","Status: 403 Forbidden - HSS User Unknown | " I don't know if any information or configuration details are missing but I can attach it at any time without any problem. Regards and thank you very much in advance, Javier Espinosa.

2 3

Kamailio Process CPU 100%
by M. Salman 05 Aug '19

05 Aug '19

Hi Team, Facing 100% Kamilio CPU issue, here are the details, any pointers/hints would help. Thanks *Scenario:* Random Kamailio CPU shoot, restart fixes it. Happens once or twice in a day. It's a pre-production server with around 30 extensions, not much load. For now, it has 2 CPUs and 4 G memory. *[Info]* uname -a Linux ip-X-aws #46-Ubuntu SMP Thu Jul 4 13:38:28 UTC 2019 x86_64 x86_64 x86_64 GNU/Linux kamailio -V version: kamailio 5.2.3 (x86_64/linux) c36229 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c36229 *(gdb) bt full* #0 0x00007f0069d97aab in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #1 0x00007f0069d7d6bd in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #2 0x00007f0069d657c7 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #3 0x00007f0069d3ebce in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #4 0x00007f0069d3fc4c in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #5 0x00007f0069d33edc in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #6 0x00007f0069d34fa8 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #7 0x00007f0069d36d71 in mysql_real_query () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 No symbol table info available. #8 0x00007f006a31e4b4 in db_mysql_submit_query (h=0x7f006a6cdb20, s=0x7f0069af92d0 <sql_str>) at km_dbase.c:116 t = 1564528172 i = 0 code = 0 _func_ = "db_mysql_submit_query" #9 0x00007f00698e8fb3 in db_do_submit_query (h=0x7f006a6cdb20, query=0x7f0069af92d0 <sql_str>, submit_query=0x7f006a31dd3c <db_mysql_submit_query>) at db_query.c:58 ret = 10 tvb = {tv_sec = 0, tv_usec = 0} tve = {tv_sec = 0, tv_usec = 2077076196242373120} tz = {tz_minuteswest = 0, tz_dsttime = 0} tdiff = 0 _func_ = "db_do_submit_query" #10 0x00007f00698ec1dc in db_do_delete (h=0x7f006a6cdb20, k=0x7ffcdb440fa0, o=0x0, v=0x7ffcdb440fb0, _n=2, val2str=0x7f006a340429 <db_mysql_val2str>, submit_query=0x7f006a31dd3c <db_mysql_submit_query>) at db_query.c:299 off = 68 ret = 36 ---Type <return> to continue, or q <return> to quit--- _func_ = "db_do_delete" #11 0x00007f006a322f6a in db_mysql_delete (h=0x7f006a6cdb20, k=0x7ffcdb440fa0, o=0x0, v=0x7ffcdb440fb0, _n=2) at km_dbase.c:515 No locals. #12 0x00007f0065b6da70 in remove_dialog_from_db (cell=0x7f005c5ed3d8) at dlg_db_handler.c:722 values = {{type = DB1_INT, nul = 0, free = 477963473, val = {int_val = 1460, ll_val = 140720308487604, double_val = 6.9525070095907518e-310, time_val = 140720308487604, string_val = 0x7ffc000005b4 <error: Cannot access memory at address 0x7ffc000005b4>, str_val = { s = 0x7ffc000005b4 <error: Cannot access memory at address 0x7ffc000005b4>, len = 0}, blob_val = { s = 0x7ffc000005b4 <error: Cannot access memory at address 0x7ffc000005b4>, len = 0}, bitmap_val = 1460, uint_val = 1460, ull_val = 140720308487604}}, {type = DB1_INT, nul = 0, free = 0, val = {int_val = 8020, ll_val = 139637976735572, double_val = 6.899032715982534e-310, time_val = 139637976735572, string_val = 0x7f0000001f54 <error: Cannot access memory at address 0x7f0000001f54>, str_val = { s = 0x7f0000001f54 <error: Cannot access memory at address 0x7f0000001f54>, len = 0}, blob_val = { s = 0x7f0000001f54 <error: Cannot access memory at address 0x7f0000001f54>, len = 0}, bitmap_val = 8020, uint_val = 8020, ull_val = 139637976735572}}} match_keys = {0x7f0065dfc6c0 <h_entry_column>, 0x7f0065dfc6b0 <h_id_column>} vars_match_keys = {0x7f0065dfc7f0 <vars_h_entry_column>, 0x7f0065dfc7e0 <vars_h_id_column>} _func_ = "remove_dialog_from_db" #13 0x00007f0065b49375 in destroy_dlg (dlg=0x7f005c5ed3d8) at dlg_hash.c:371 ret = 1 var = 0x7f0065b45442 <futex_get+40> _func_ = "destroy_dlg" #14 0x00007f0065b50ca0 in dlg_unref_helper (dlg=0x7f005c5ed3d8, cnt=2, fname=0x7f0065be1eff "dlg_handlers.c", fline=1182) at dlg_hash.c:961 d_entry = 0x7f005c45b4c0 _func_ = "dlg_unref_helper" #15 0x00007f0065b7f708 in unref_dlg_from_cb (t=0x7f005c5b5dc0, type=131072, param=0x7ffcdb4412a0) at dlg_handlers.c:1182 dlg = 0x7f005c5ed3d8 iuid = 0x7f005c568c38 #16 0x00007f0068295b05 in run_trans_callbacks_internal (cb_lst=0x7f005c5b5e38, type=131072, trans=0x7f005c5b5dc0, params=0x7ffcdb4412a0) at t_hooks.c:260 cbp = 0x7f005c606118 backup_from = 0x5569cc3da8b0 <def_list+16> backup_to = 0x5569cc3da8b8 <def_list+24> ---Type <return> to continue, or q <return> to quit--- backup_dom_from = 0x5569cc3da8c0 <def_list+32> backup_dom_to = 0x5569cc3da8c8 <def_list+40> backup_uri_from = 0x5569cc3da8a0 <def_list> backup_uri_to = 0x5569cc3da8a8 <def_list+8> backup_xavps = 0x5569cc3da2c0 <_xavp_list_head> _func_ = "run_trans_callbacks_internal" #17 0x00007f0068295c2c in run_trans_callbacks (type=131072, trans=0x7f005c5b5dc0, req=0x0, rpl=0x0, code=0) at t_hooks.c:287 params = {req = 0x0, rpl = 0x0, param = 0x7f005c606128, code = 0, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = { s = 0x0, len = 0}} #18 0x00007f00681cbf45 in free_cell_helper (dead_cell=0x7f005c5b5dc0, silent=0, fname=0x7f00682de882 "timer.c", fline=689) at h_table.c:166 b = 0x7f005c5b5dc0 "" i = 1 rpl = 0x7ffcdb4413a0 tt = 0x0 foo = 0x7ffcdb441390 cbs = 0x7f005c378700 cbs_tmp = 0x1681ca80f _func_ = "free_cell_helper" #19 0x00007f0068283bf6 in wait_handler (ti=855479619, wait_tl=0x7f005c5b5e48, data=0x7f005c5b5dc0) at timer.c:689 p_cell = 0x7f005c5b5dc0 ret = 1 unlinked = 0 rcount = 1 _func_ = "wait_handler" #20 0x00005569cbdb0661 in timer_list_expire (t=855479619, h=0x7f005c238af0, slow_l=0x7f005c23ae08, slow_mark=5646) at core/timer.c:874 tl = 0x7f005c5b5e48 ret = 0 #21 0x00005569cbdb0b09 in timer_handler () at core/timer.c:939 saved_ticks = 855479619 run_slow_timer = 0 i = 526 _func_ = "timer_handler" #22 0x00005569cbdb0fd2 in timer_main () at core/timer.c:978 ---Type <return> to continue, or q <return> to quit--- No locals. #23 0x00005569cbcf1382 in main_loop () at main.c:1727 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=172.31.2.33:6000 ( 13.54.252.29:6000)\000\034\004\000\000\000\000\000\000\000\321$}\034\000\000\000\000P0\316\313iU\000\000 \031D\333\374\177", '\000' <repeats 18 times>, "`\026D\333\374\177\000\000\360S\361\313iU\000" nrprocs = 8 woneinit = 1 _func_ = "main_loop" #24 0x00005569cbcf8a7b in main (argc=13, argv=0x7ffcdb441928) at main.c:2696 cfg_stream = 0x5569cd7c7260 c = -1 r = 0 tmp = 0x7ffcdb441f39 "" tmp_len = -616294480 port = 32764 proto = 3 options = 0x5569cc074190 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 3114863458 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7f006af01787 p = 0x9c0 <error: Cannot access memory at address 0x9c0> st = {st_dev = 23, st_ino = 456, st_nlink = 2, st_mode = 16877, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 60, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1564229247, tv_nsec = 809557407}, st_mtim = {tv_sec = 1564229229, tv_nsec = 217250917}, st_ctim = {tv_sec = 1564229279, tv_nsec = 666090748}, __glibc_reserved = {0, 0, 0}} _func_ = "main" (gdb) -- Regards M. Salman

3 4

alias="server" from db?
by David Villasmil 05 Aug '19

05 Aug '19

Hello guys, I can't find a module that does this, i want to load my hostname aliases from db, instead of alias="host1:5060" alias="host2:5060" alias="host3:5060" I assume it is possible, but how? first time i need to implement this Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 6

Re: [SR-Users] Kamailio v5.2 kamctl alias add problem
by Igor Potjevlesch 04 Aug '19

04 Aug '19

Hi kamailio Users, I meet a problem with the " kamctl alias add" command in the 5.2 version, I saw that the module used to add an alias had changed, and now the command result with an error : [kamailio]# kamctl alias add XXXX@XXXX sip:XXXXX@XXXX { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters or wrong format" }, "id": XXXXX } So in the version 5.2, the table aliasdb replace the table alias ? We are using the table alias for the incomming call, and we implemented scripts to add with the command kamctl alias add the information needed in the tavble alias to route the call. But since the version 5.2 we need to add an argument "path" in this command, and we don't know how to put it at "NULL" or how we can use this argument ? Can you help us with this? Bests Regards Igor Le ven. 5 juil. 2019 à 16:57, Igor Potjevlesch <igor.potjevlesch(a)gmail.com> a écrit : > Hi Daniel-Constantin, > > So in the version 5.2, the table aliasdb replace the table alias ? > We are using the table alias for the incomming call, and we implemented > scripts to add with the command kamctl alias add the information needed in > the tavble alias to route the call. > But since the version 5.2 we need to add an argument "path" in this > command, and we don't know how to put it at "NULL" or how we can use this > argument ? > > Can you help us with this? > > Bests Regards > >>

3 3

Reverse Proxy Asterisk - SIP Provider
by Mihai Cezar 04 Aug '19

04 Aug '19

Hi all, I've tried to create a reverse proxy to forward incoming request that came from SIP provider to Asterisk PBX and forward the requests from asterisk to kamailio then sip provider. What i get is that I see the invite, but is like no ACK. Thanks in advance. M kamailio.cfg: #!KAMAILIO # ####### Defined Values ######### # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=3 log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " children=1 server_id = 10 xavp_via_params = "via" disable_tcp=yes auto_aliases=no listen=udp:0.0.0.0:5060 ####### Modules Section ######## loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 0) modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) ####### Routing Logic ######## request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE|REFER")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); sl_send_reply("100","Trying"); if ($si == "172.16.16.1") { sl_send_reply("183","Incoming session from Avoxi"); rewritehost("10.1.1.10"); #exit; } else if ($si == "10.1.1.10"){ # receiving response from client sl_send_reply("183","Outgoing session to Avoxi"); #rewritehost("172.16.16.1"); drop; exit; } else { sl_send_reply("500","No configured IP!"); drop; exit; } } if ($rU==$null) { sl_send_reply("484","Address Incomplete"); exit; } # received from main server - send to client and add via tokens for anycast handling via_add_srvid("1"); $xavp(via=>node) = "10.1.1.4"; via_add_xavp_params("1"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } if ($si == "10.1.1.4") { xlog("L_WARN", "$ci|end|dropping message"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("400","Loop detected"); exit; } # TM manage for outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # TM manage for incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } # TM manage for failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } asterisk - sip.conf [opensips-trunk](sip-provider) fromdomain=10.1.1.10 host=10.1.1.4 context=from-trunk type=friend insecure=invite,port trunk=yes

4 9

Kamailio OPTIONS Round-Trip
by Eugeniu Istrati 04 Aug '19

04 Aug '19

I'm interested in this feature too, would it be available in the 5.3 release? Thank you. -- Eugeniu Istrati

2 1

location table
by Maryam Baghdadi 03 Aug '19

03 Aug '19

Hello How can I save subscribers information in location table? why after calling save("location") in configuration file , still location table is empty?

2 1

(no subject)
by jay binks 02 Aug '19

02 Aug '19

Hey all, looking for a little help. I've been tracking an issue, where a number of SIP Messages ( typically INVITE ) are sent, but Kamailio only see one of them ( the last one). I have verified that the messages are definitely received on the host, as verified with TCPDUMP, but the INVITES never hit the request route block. ( the very first thing is to log the message ) What i've managed to find is, that if I increase worker children, the problem goes away. So I've got a problem, I think I've found the solution, but what I'm struggling with is how to monitor better, so I can be alerted to this in the future. I started by looking at "kamctl stats | grep drop" "core:drop_replies = 14", "core:drop_requests = 10577" however drop_requests seems to include explicit drops in my config. which I do for many reasons, but mainly bad UA or blocked IP's, so this dosnt seem to be what im after. I then went digging in the code ( often the best way to find things ). in receive.c receive_msg() we find some conditions, where we drop the packet ( before calling request route ). right now im looking around line 251 at LM_DBG("dropping the received message\n"); goto error00; Following error00 I see it calls STATS_RX_DROPS which increments stats->received_drops. This is where it gets fun, I cant find where received_drops is ever returned in the stats output. I can see where -SIGUSR1 will trigger this to be dumped, if Kamailio is compiled with that option ( dosnt seem the debian packages are ). however, what im seeking clarification on is, does this get reported in "kamctl stats" Am I just going crazy or is something not quite right here? TLDNR; how do I monitor for Kamailio not having enough worker threads to process incoming messages. FYI I'm seeing about 50Mbps of incoming SIP on this box, and I do perform a decent amount of work on some message types. -- Sincerely Jay

5 7

Deleting extra location attribute(location_attrs table) when contact is expired
by Pintu Lohar 31 Jul '19

31 Jul '19

Hi All, Regarding modparam("tm|usrloc", "xavp_contact", "ulattrs") #ref : https://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.xa… . There is some requirement where we need to save some extra location attributes to location_attrs <https://kamailio.org/docs/db-tables/kamailio-db-4.3.x.html#gen-db-location-…> tables $xavp(ulattrs=>deviceIdentify)= "Some str values" and calling save(“location”) , It works well and saved location attribute table but But when contact expired callback is received, udomain_contact_expired_db() does not it delete the extra location attribute from database but deleting the record from location table only ? If not, is there any recommended approach to delete the extra location attributes from the database? or am I missing something to set? Alternate I was thinking could be running timer from Kamailio config and check last_modified time and delete those(based on our UAC expiry setting), is it the right approach. Kamailio Version: 5.2.0. Thanks for your valuable suggestion in advance. Regards Pintu Lohar

2 10

Register without user create
by José Antonio Gutiérrez Delgado 29 Jul '19

29 Jul '19

Hello, I would like to know if it is possible to enable registration for users that have not been created with kamctl. For example, if I can register a user with a random name without a password, or obtain from another database other than kamailio; I would also like it to be allowed at the same time as there are users with password registered with kamctl. Thanks you, José Antonio Gutiérrez Delgado Responsable técnico Oficina: (+34) 923040031 <http://www.arsoft-company.com> *De conformidad a lo establecido en el REGLAMENTO (UE) 2016/679 DEL PARLAMENTO EUROPEO Y DEL CONSEJO de 27 de abril de 2016 relativo a la protección de las personas físicas en lo que respecta al tratamiento de datos personales y a la libre circulación de estos datos y por el que se deroga la Directiva 95/46/CE (Reglamento general de protección de datos), le informamos que todos los datos pasaran a formar parte de un fichero automatizado cuyo responsable es AUGMENTED REALITY SOFTWARE S.L siendo la única finalidad de dicho fichero la gestión de carácter comercial y el posible envío de comunicaciones comerciales sobre nuestros productos. Si lo desea puede ejercer sus derechos de acceso, rectificación, cancelación y oposición de sus datos en C/ Duero 12, Parque Científico, ARSOFT, 37185 o enviando un mensaje electrónico a info(a)arsoft-company.com <info(a)arsoft-company.com> indicando en el asunto el derecho que desea ejercitar.Este mensaje y sus archivos adjuntos van dirigidos exclusivamente a su destinatario, pudiendo contener información confidencial sometida a secreto profesional. No está permitida su reproducción o distribución sin nuestra autorización expresa. Si usted no es el destinatario final por favor elimínelo e infórmenos por esta vía.*

3 4

Dialog missing when BYE is received from trunk
by Andrew White 28 Jul '19

28 Jul '19

Hi all, I’ve got a call flow where PBX makes a call via Kamailio, which uses the UAC modules to make the call to the trunk. This works great, however I’ve noticed recently that when the BYE comes from the trunk, Kamailio doesn’t recognise the dialog and throws back a 404. Due to this, the PBX never knows the session has ended, and keeps its session open. Here’s the flow from Wireshark: https://i.imgur.com/BxPRrXP.png <https://i.imgur.com/BxPRrXP.png> As you can see, the BYE is received from the trunk, and Kamailio throws a 404. I checked with an rpc dlg.list and at this point the dialog had gone. 30 seconds later, I threw a BYE from the PBX, and because the dialog no longer existed, it got a 481 back. The INVITE to the trunk has the following Record-Route header: Record-Route: <sip:kamailio.public-ip;lr;ftag=mjKtcKaBteQ4Q;vst=AAAAAAYFBQUHAwYHBgRzMwIRGRMcAgJBWBEVCwAbBhtADxBjdGVkLmNvbS5hdQ--;did=624.b2e1> When the first BYE comes in, it still has same did in the Route header. Below I’ve put the output from the dialog list rpc call (before the first BYE) and the relevant log lines. { h_entry: 1062 h_id: 7723 ref: 2 call-id: a9d32620-2b68-1238-048a-0cc47a18705a from_uri: sip:61400123123@pbx.local:5060 to_uri: sip:61400123123@10.0.0.2 state: 4 start_ts: 1564269942 init_ts: 1564269929 end_ts: 0 timeout: 1564313142 lifetime: 43200 dflags: 643 sflags: 0 iflags: 16 caller: { tag: mjKtcKaBteQ4Q contact: sip:pbx.local cseq: 7581172 route_set: socket: udp:10.0.0.2:5060 } callee: { tag: SD5ptp899-1177933201-1564269931305 contact: sip:SDd3mt8-nn9jnirnmqcrvfmvkpm0hmhvgggjgv2io788of70s84u9@trunk.remote:5060;transport=udp cseq: 0 route_set: socket: udp:10.0.0.2:5060 } profiles: { } variables: { { cseq_diff: 1 } } } Jul 28 09:27:30 ip-10-0-0-2 /usr/local/sbin/kamailio[11061]: NOTICE: {2 450917085 BYE a9d32620-2b68-1238-048a-0cc47a18705a} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1564270050;method=BYE;from_tag=SD5ptp899-1177933201-1564269931305;to_tag=mjKtcKaBteQ4Q;call_id=a9d32620-2b68-1238-048a-0cc47a18705a;code=404;reason=Not here;src_user=0400123123;src_domain=trunk.remote;src_ip=trunk.remote;dst_ouser=0400123123;dst_user=0400123123;dst_domain=kamailio.public-ip I’m assuming the 404 is because the Contact header I sent from Kamailio -> Trunk is rewritten, that Contact is Kamailio’s public contact IP. However the dialog has the caller contact within it referencing back to the local PBX, so I would think when Kamailio gets this dialog ID back with a BYE it would relay it back to the original contact, wouldn’t it? Thanks! Andrew

2 3

IMS_ISC technical specification
by Ali Shirvani 26 Jul '19

26 Jul '19

Hi all, I read some portion of imc_isc_mod.c, specially isc_from_as function. In that function there is some function call that searching for some marks in the SIP message. Also I read section 5.7.3 from TS 24.229 but I didn't find the marking mechanism that used in isc_from_as function. Would you please point me to the technical specification that implemented in IMS_ISC module? Regards, Ali

3 4

Concatenating strings in $hdr
by David Cunningham 26 Jul '19

26 Jul '19

Hello, I'm having an issue where this doesn't work: #!trydef PRODUCT "Foo" if( $hdr("X-" + PRODUCT + "-Transport") != "" ) { add_uri_param( $hdr("X-" + PRODUCT + "-Transport") ); } It seems Kamailio doesn't like the concatenation within the $hdr(). Concatenation does work in other places though: append_hf( "X-" + PRODUCT + "-Transport: tcp\r\n" ); Can anyone advise how to make the concatenation within $hdr() work? Thanking you in advance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

IM Chat facility in Kamailio
by Gaurav Bmotra 25 Jul '19

25 Jul '19

hi i m using kamailio 5.1.2 , on Ubuntu 18. LTS existing setup is working fine with audio and video call ,,, ut now i wan to add chat feature in it ... so plz help me how can i implement chat IM in to it thanks -- *Regards:* Gaurav Kumar

2 1

Malformed Route Header
by Duarte Rocha 25 Jul '19

25 Jul '19

Greetings, When a in-dialog request has Route headers, Kamailio will remove the Route Header with it's address when routing the packet. This works perfectly for me in every situation. However, in one of the calls I found a weird behaviour in the removal. So, this is the scenario : Kamailio has two IP's 1.1.1.209 and 1.1.1.205. This is BYE request received in the on 1.1.1.209 The header Route looks like this when it arrives at 1.1.1.209 : Route : <sip:1.1.1.209;r2=on;lr;ftag=5AE0303037363132017D0E65;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQECA3ByAwMcHwIdHQsWHwUMOA--;vst=AAAAABgHDQgBAwsIBQECdnIDAxwfAh0aBB4cAgU5;did=8f5.67c2>,<sip:1.1.1.205;r2=on;lr;ftag=5AE0303037363132017D0E65;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQECA3ByAwMcHwIdHQsWHwUMOA--;vst=AAAAABgHDQgBAwsIBQECdnIDAxwfAh0aBB4cAgU5;did=8f5.67c2> It reaches 1.1.1.205 like this : Route : As you can see, the header is empty and it should contain the 1.1.1.205 URI. Is the header Router malformatted and creating this error ? Best Regards, Duarte Rocha

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Re: [SR-Users] [EXT] Re: P-CSCF and I-CSCF ports -RESOLVED
by Woscek, Martin W. 24 Jul '19

24 Jul '19

Hi Ali, Looking through the source code, its not that the port for I-CSCF is hard coded to 5060, its just that it defaults to 5060 if the following is set in the Kamailio.cfg file: dns_try_naptr=no which we had left it disabled. Well I assumed I had all the DNS worked out, I had NAPTR records set for the P,I and S-CSCF. Looking though the pcscf.cfg there is a setting for NETWORKNAME which is set to ims522.ims.org, and is a separate FQDN from the P,I,S-cscf FQDN’s. I always kept the NAPTR rec for ims522 as the same port of the P-CSCF (pcscf522) in DNS. Two things I needed to fix: 1. Enable the DNS NAPTR lookups: dns_try_naptr=yes 2. Modify the NAPR for ims522 to be that of 4060 Thanks to you and Amar for the pointing to DNS. _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Ali Shirvani Sent: Tuesday, July 23, 2019 1:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] P-CSCF and I-CSCF ports Hi Martin, As Amar pointed out it may some DNS configuration. You can check the DNS entry of I-CSCF with: $ dig -t srv <FQDN of I-CSCF> Regards, Ali On Tue, Jul 23, 2019 at 7:15 PM Amar Tinawi <amar.tinawi(a)gmail.com<mailto:amar.tinawi@gmail.com>> wrote: Hello Martin I think you are missing the DNS part. On Tue, Jul 23, 2019, 5:30 PM Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> wrote: Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin _______________________________________________ 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 _______________________________________________ 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

1 0

kamailio 4.4.4 fills IPv6 contact address when sending packet over IPv4
by Pali Rohár 24 Jul '19

24 Jul '19

Hi! I detected strange problem that sip.linphone.org refuse to accept presence information re-transmitted by kamailio 4.4.4 server. I debug this problem with tcpdump and I found out that problem is in kamailio which fills IPv6 address into UDP datagram and that datagram is sent via IPv4 socket to IPv4 address, to sip.linphone.org server. And sip.linphone.org server does not have IPv6 connectivity, so correctly return over IPv4 to sender just "400 Bad Contact Header" error. On my server is running kamailio 4.4.4 from Debian Stretch and I can 100% reproduce this problem against public sip.linphone.org server. My server has both IPv4 and IPv6 connectivity and kamailio is listening for both IPv4 and IPv6 connections. So why is kamailio sending IPv6 address over IPv4 and therefore makes it impossible to communicate with non-IPv6 enabled servers? Looks like a problem with choosing default/correct socket for Contact header. And how to fix this problem? Can you help me? I would like to have working interconnection with linphone servers. Just to note I'm seeing this problem only for presence information packets. Other requests, like INVITE or MESSAGE seems to work. Below is relevant tcpdump output. Some parts were replaced by {VAR}. PS: I'm not subscribed to list, so please CC my address when sending reply. Thank you! 17:22:58.121719 IP (tos 0x10, ttl 64, id 21629, offset 0, flags [none], proto UDP (17), length 1266) {MY_IPV4_ADDRESS}.5060 > 91.121.209.194.5060: [bad udp cksum 0xa099 -> 0x9825!] SIP, length: 1238 NOTIFY sip:{REMOTE_NAME}@{REMOTE_USER_IPV4_ADDRESS}:5060;registering_acc=sip_linphone_org SIP/2.0 Via: SIP/2.0/UDP {MY_IPV4_ADDRESS};branch=z9hG4bK2b55.88f93c20000000000000000000000000.0 To: <sip:{REMOTE_NAME}@sip.linphone.org>;tag=75559182 From: <sip:{MY_SIP_URI}>;tag=97d8e785fdf42bf9622a64c13c504961-2708 CSeq: 2 NOTIFY Call-ID: 26cf9d5c019af2dc3302b770887bcc2e@0:0:0:0:0:0:0:0 Route: <sip:91.121.209.194:5060;lr> Content-Length: 597 User-Agent: kamailio (4.4.4 (x86_64/linux)) Max-Forwards: 70 Event: presence Contact: <sip:{MY_IPV6_ADDRESS}:5060;transport=udp> Subscription-State: active;expires=3600 Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="Pali <sip:{MY_SIP_URI}>"> <tuple id="sg89ae"> <status><basic>open</basic></status> <contact priority="0.8">Pali <sip:{MY_SIP_URI}></contact> </tuple> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="TA0C538B2"> <status> <basic>closed</basic> </status> <contact priority="1">sip:{MY_SIP_URI}</contact> <timestamp>2019-04-19T17:20:36+02:00</timestamp> </tuple> </presence> 17:22:58.151188 IP (tos 0x0, ttl 52, id 22949, offset 0, flags [none], proto UDP (17), length 373) 91.121.209.194.5060 > {MY_IPV4_ADDRESS}.5060: [udp sum ok] SIP, length: 345 SIP/2.0 400 Bad Contact Header Via: SIP/2.0/UDP {MY_IPV4_ADDRESS};branch=z9hG4bK2b55.88f93c20000000000000000000000000.0;rport=5060 From: <sip:{MY_SIP_URI}>;tag=97d8e785fdf42bf9622a64c13c504961-2708 To: <sip:{REMOTE_NAME}@sip.linphone.org>;tag=75559182 Call-ID: 26cf9d5c019af2dc3302b770887bcc2e@0:0:0:0:0:0:0:0 CSeq: 2 NOTIFY Content-Length: 0 -- Pali Rohár pali.rohar(a)gmail.com

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How to hook to Register And Unregister event in Kamailio
by Anuran Barman 24 Jul '19

24 Jul '19

Hi, I am integrating Kamailio into my application. I want to hook to the successful REGISTER and unregister event into Kamailio into my application. For now, I am able to hook into INVITE event and can hit my server to send a email to the callee user that user X is calling you. The way I am doing is by this if (is_method("INVITE")) { xlog("LOG_LOCAL3","L_INFO","invite came ($fU) ($tU)"); $var(res) = http_connect("sipnodejsserver", "/","text/plain","src_user:$fU,dst_user:$tU" ,"$avp(route)"); xlog("LOG_LOCAL3","L_INFO","request sent $avp(route) $var(res)"); setflag(FLT_ACC); # do accounting } I need to show into my application that their SIP phone is online or not. How can I hook into this?

4 24

Re: [SR-Users] [EXT] Re: P-CSCF and I-CSCF ports
by Woscek, Martin W. 23 Jul '19

23 Jul '19

DNS is fine. I flip the NAPTR records when I flip back and forth the 4060/5060 btwn the P and I. Martin Woscek The MITRE Corp. mwoscek(a)mitre.org<mailto:mwoscek@mitre.org> Phone: 703-983-2650 FAX: 703-983-7142 From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Ali Shirvani Sent: Tuesday, July 23, 2019 1:40 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] Re: [SR-Users] P-CSCF and I-CSCF ports Hi Martin, As Amar pointed out it may some DNS configuration. You can check the DNS entry of I-CSCF with: $ dig -t srv <FQDN of I-CSCF> Regards, Ali On Tue, Jul 23, 2019 at 7:15 PM Amar Tinawi <amar.tinawi(a)gmail.com<mailto:amar.tinawi@gmail.com>> wrote: Hello Martin I think you are missing the DNS part. On Tue, Jul 23, 2019, 5:30 PM Woscek, Martin W. <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> wrote: Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin _______________________________________________ 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 _______________________________________________ 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

1 0

P-CSCF and I-CSCF ports
by Woscek, Martin W. 23 Jul '19

23 Jul '19

Hi IMS users, I can configure the P-CSCF to use 5060, and the I-CSCF to use 4060 and both come up listening to their respective ports when each module is started. But the P-CSCF is hardcoded to use 5060 for the I-CSCF. Before I make code changes to the executable: Is there some additional configuration to allow for me to switch ports? Why is Kamailio IMS hardcoded and why not enforce it at the .cfg load time? Thanks, Martin

3 2

Debugging kamailio module
by Ali Shirvani 23 Jul '19

23 Jul '19

Hi all, I want to trace function calls in a kamailio module, e.g. ims_isc module. I installed kamailio-dbg package and started the kamailio with gdb: `gdb kamailio` I also set break point on target function, but it seems gdb doesn't stop on that break point. How should I start gdb and set break points? it seems gdb couldn't handle kamailio child processes that forked in start. Regards, Ali

2 2

unsubscribe
by Carlos Cruz 23 Jul '19

23 Jul '19

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2 1

unsubscribe
by Carlos Cruz 23 Jul '19

23 Jul '19

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3 2

Two GWs with the same IP
by Duarte Rocha 22 Jul '19

22 Jul '19

Greetings, I'm building some testing procedures for my Kamailio proxy. In order to do that i have a SipTester (Acting as Calling Number) from where i want to originate calls to Kamailio. After Kamailio processes the call, i want it to route it to SipTester again so it can act as Called Number. This should work with only one SipTester in one machine, so i created two SipTester GW's on Kamailio, belonging to different users and GW groups. Also, the two GWs use different Kamailio addresses to communicate with the proxy. However, in this combination, when routing i get this error " ERROR: dialog [dlg_handlers.c:666]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/". If i change the destination or the origin of the INVITE the calls work without a problem. Does this happen because both GW's have the same address? Best Regards

2 1

Reading presence state
by Gertjan Wolzak 19 Jul '19

19 Jul '19

Hello Kamailions, I have been able to get the presence configured, subscribers see the status of their subscriptions, this is working fine, for internal, incoming external and external calls. This was solved by enabling use_pubruri_avps in the pua_dialoginfo module. Now I want to be able to route calls based on the presence state. Using the presentity table could be used by doing a query based on the userpart of the ruri, which works fine when the call is still active, but it can take a while for the record to be removed from the presentity table, as that is dependend on the expires value. So when a query is made after a call is finished, a record would still be present in the presentity table till clean_period timer has hit again and the expires value is expired. So I should get the status from memory, but I have not been able to find the information on how to do that. Can someone enlighten me on where to find that information? Rgds, Gertjan

2 4

I need to hold Leg A of a call for a random amount of time
by Gaurav Bmotra 19 Jul '19

19 Jul '19

hi i m using kamailio where i m tring to hold the call bitween switch A and Switch B PSTN -> Switch A -> SWITCHB -> SwitchC -> DESTINATION When PSTN hangs up, I need the call to stay up between SwitchA and B for an arbitrary number of seconds or minute plz help thank you -- *Regards:* Gaurav Kumar

2 1

Kamailio 5.2.3 failed to start due to incompatible openssl version
by Andrew Chen 18 Jul '19

18 Jul '19

Hey support, Today I ran into this error on my latest 5.2.3 Kamailio box: Jul 18 15:05:28 sjomaintpsg51 /usr/sbin/kamailio[5525]: CRITICAL: tls [tls_init.c:671]: init_tls_h(): installed openssl library version is too different from the library the kamailio tls module was compiled with: installed “OpenSSL 1.1.1 11 Sep 2018” (0x1010100f), compiled “OpenSSL 1.1.0g 2 Nov 2017" (0x1010007f).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) There's not support for openssl1.1.1 ? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *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 3

kamailio + asterisk integration RTP flow question
by Satish Patel 18 Jul '19

18 Jul '19

I am configuring kamailio as registar and asterisk for media services. in kamailio + asterisk integration how two SIP clients RTP traffic will flow, are they go via Asterisk or directly from kamailio ?

3 2

SIP-DTLS support
by Martin Meadows 17 Jul '19

17 Jul '19

Hello, Has there been any movement on the support of SIP-DTLS (datagram-TLS) within Kamailio since: * sr-users:2012-October:075185.html <https://lists.kamailio.org/pipermail/sr-users/2012-October/075185.html> I'm interested to see if Kamailio can proxy 5061/udp to a FreeSWITCH 5060/udp. In general, I have been unable to find any material on implementations of: * draft-jennings-sip-dtls-05.html <https://tools.ietf.org/id/draft-jennings-sip-dtls-05.html> Cheers - Martin

4 6

opt_reply(): called for non-OPTIONS request
by Alexandru Covalschi 17 Jul '19

17 Jul '19

Hello list, if KSR.is_OPTIONS() and KSR.is_myself_ruri(): KSR.x.modf("options_reply"); exit(); with a valid URI (no username) gives the subj output (opt_reply(): called for non-OPTIONS request) How could that be? Python KEMI, 5.2.3, docker. ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

2 5

Kamailio IMS to IP PBX - how?
by David Dean 16 Jul '19

16 Jul '19

I'm trying to connect a VoLTE smartphone to an IP PBX using Kamailio IMS. I've managed to get the smartphone to connect to Kamailio IMS and make calls with our devices registered with Kamailio. The problem is that I need the smartphones to register with the IP PBX which is a VigorBX 2000n. This is so they can use an internal extension number and make calls to other local extensions with hard wired desk phones. Should I be using Kamailio IMS to "proxy" the smartphone SIP connection to the VigorBX PBX? Or would it be better to connect the smartphone directly to the VigorBX using SIP? If so, how? I've read absolutely every how-to document I could lay my hands on but none of them describe integrating IMS with a PBX. Thanks for any help / advice you can offer. Dave

1 0

KEMI beginner question on alpine Linux
by Karsten Horsmann 16 Jul '19

16 Jul '19

Hi List, I play around with alpine Linux 3.10.x and Kamailio 5.2.x and I search for app_python3 and KEMI support. But I don't found it. AFAIK if I want to use KEMI with Python3 I need of course Kamailio and app_python3 (maybe something else?). Is the package maintainer on this list? Cheers Karsten

4 4

How to get total number of dialogs in profile with value
by Nathan Angelacos 15 Jul '19

15 Jul '19

In kamailio < 5.1.x it used to be possible to get total number of items in a dialog_with_profile by skipping the "value" paramter, such as: modparam ( "dialog", "profiles_with_value", "outbound" ) . . . get_profile_size ( "outbound", "$var(myvar)" ); However, 5.1.x and 5.2.x report: ERROR: dialog [dialog.c:912]: w_get_profile_size_helper(): invalid value parameter The documentation ( https://kamailio.org/docs/modules/5.2.x/modules/dialog.html#dialog.f.get_pr… ) Indicates the "value" paramter is still optional. I'm sure I must be missing something obvious... Any ideas? Thanks!

2 3

UDP fragmentation issues
by Gerry | x164 15 Jul '19

15 Jul '19

Hi, I have setup Kamailio on Centos 7 as a proxy in front of Asterisk in the same datacenter, using the dispatcher module. I have noted that large INVITEs sent by Kamailio do not arrive at the Asterisk box. It seems the network is dropping fragmented UDP traffic. This does not happen however when I contact the Asterisk box directly from outside the datacenter. Then large INVITES are arriving are Asterisk. So the problem is either in Kamailio or in the datacenter. I have analysed the problem with ngrep and tcpdump/wireshark. In normal mode (udp4_raw=no setting) Kamailio is producing UDP packets above 1500 bytes and leaves all fragmentation to the operating system. Enabling raw udp in Kamailio and lowering the MTU in the server didnt help. I even disabled offloading of UDP fragmentation to the NIC to get to the bottom of this. However nothing helps. Only forcing TCP in the dispatcher works when having large INVITES going to Asterisk. In any case, it appears that there is no guarantee that fragmented UDP arrives at all. As far as I am aware, the recommendation is to use TCP if messages are bigger than 1400 bytes. If so, how can I force Kamailio to send messages over 1400 bytes only in TCP mode? Or is there some other way how to deal with this? Thanks for any hints. Gerry

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Kamailio Asterisk integration with docker DNS Problem
by Amir 15 Jul '19

15 Jul '19

Hi I have created a kamailio container with docker and two asterisk container And this is my dispatcher list: *1 sip:asterisk:50601 sip:asterisk2:5060* and this is my both asterisk SIP.conf *[Kamailio]host=kamailioport=5060insecure=invitetype=friendcontext=from-internal* The problem is that I use User-defined networks which has its own DNS when a container starts a dns record would be set If I start a kamailio first then start two asterisks the asterisk works fine because the name "kamailio" has been set in docker dns but I should restart the kamailio because at startup It could not resolve astersik dns records after the everything works fine but if the any of the asterisk instances crash/stop kamailio can not detect that because a dns record would not be available: *17(22) ERROR: <core> [core/resolve.c:1684]: sip_hostport2su(): could not resolve hostname: "asterisk2"17(22) ERROR: tm [ut.h:309]: uri2dst2(): failed to resolve "asterisk2"17(22) ERROR: tm [uac.c:452]: t_uac_prepare(): no socket found17(22) ERROR: dispatcher [dispatch.c:3110]: ds_ping_set(): unable to ping [sip:asterisk2:5060]* It is kind of loop , I think maybe I can fix asterisk problem with asterisk realtime and set the sip.conf in database. anyone has any suggestion for this problem?

8 14

Wrong $rm value
by Alexandru Covalschi 14 Jul '19

14 Jul '19

Hello list, I use kamailio 5.2.3 from repo's on stretch in docker. I use python KEMI + async and encountered weird behaviour: after t_continue $rm shows OPTIONS instead of REGISTER/INVITE. I wonder if that's correct behaviour, because for me it looks like a bug. Repeating is easy - just check $rm after t_continue. ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

1 2

Check if an INVITE comes from uac_reg endpoint
by Diogo Hartmann 14 Jul '19

14 Jul '19

Hello Kamailio experts, We were already able to register correctly at a SIP Provider using uacreg table. Now, we would like to validate inbound calls coming from this registered provider, sending them to a specific dispatcher group. I tried reading the uac module docs, found *uac_reg_lookup* function, but i'm not sure if that's we need. Could someone please help me to understand if UAC has a function to do that, or if we need to make some? Thanks!!

3 4

kamailio as ICSCF connection with FHoss
by Maryam Baghdadi 14 Jul '19

14 Jul '19

hello, I use Kamailio as my Icscf in IMS network . after receiving Invite Request from SCSCF to ICSCF‌, a LIR request sent from ICSCF to FHoss. User_Unknown response received in all states. I checked every thing carefully , users are registered and defined in HSS and every configuration done . What's wrong? is it possible that icscf has problem? or it can be FHoss bug?

4 14

call stuck and disconnect in kamailio
by Gaurav Bmotra 13 Jul '19

13 Jul '19

hi i m using kamailio 5.1.2 with ubuntu 18.4 LTS , and rtpproxy (Basic version: 20040107 ) we are using it for video calling everything is working fine but call get stuck after about 2 min and disconnect plz help -- *Regards:* Gaurav Kumar

2 3

Long distance rtpengine communication
by Koray Vatansever 12 Jul '19

12 Jul '19

Hi all, We have kamailio servers which communicate with the geographically distributed rtpengine servers. Sometimes udp packets are lost or delayed between kamailio and rtpengine. And this causes call drops... Since rtpengine communication is synchronized and tcp worker processes are blocked, I couldn't set higher timeout values and more retry count. Are there any suggestions to overcome this problem? Thanks, Koray

2 1

Variables for P-Called-Party-ID Header
by Benjamin Tietz 12 Jul '19

12 Jul '19

Hi, I have some INVITES where the called number is in the P-Called-Party-ID header: P-Called-Party-ID: <sip:+123456789@sip-trunk.example.com;user=phone> Is there any function or variable where the user is stored or made available? In this case +123456789 Thanks, Benjamin

2 3

SIP URI Evaluation and Forwarding
by Ronald Hartmann 12 Jul '19

12 Jul '19

Good Day List, I am very new to Kamailio and trying to understand at 10,000 feet how it may be able to be used. I was wondering if the following Scenario would be possible and if so what modules would accomplish this aspect. (Any URLs for further reading would be appreciated) Scenario: Given 1) SoftPhones will register as follows sip uri, user(a)subdomain.myvoipserver.com 2) The following users are authorized to connect (this would be stored in a config file etc): 1001,1003,1007 are valid users for apples.myvoipserver.com 1001,2213,5817 are valid users for oranges.myvoipserver.com Is it possible for Kamailio to perform the following evaluation and then take the appropriate steps: If [domain is apples.myvoipserver.com AND user portion is 1001] then Forward/Proxy the SIP/RTP packets to Asterisk server responsible for registration and audio for the domain apples.myvoipserver.com Else if [domain is oranges.myvoipserver.com AND user portion is 2213] then Forward/Proxy the SIP/RTP packets to Asterisk server responsible for registration and audio for the domain oranges.myvoipserver.com Else Silently Drop packets etc. given the above evaluation rules, a user who tries to register as 1001(a)apples.myvoipserver.com would be granted access and their SIP Packets and subsequent RTP would be forwarded to the asterisk server at apples.myvoipserver.com, where said server would take care of the Registration, and any subsequent calls, subscriptions, Options etc. 1004@ apples.myvoipserver.com Would not match the authorized list for apples.myvoipserver.com (1001,1003,1007), therefore this registration attempt would be silently dropped 2213(a)apples.myvoipserver.com would be granted access and their SIP Packets and subsequent RTP would be forwarded to the asterisk server at oranges.myvoipserver.com, where said server would take care of the Registration, and any subsequent calls, subscriptions, Options etc. Thanks for any information to assist me in understanding how this would work. ~ron

3 2

Re: [SR-Users] kamailio warning
by Henning Westerholt 10 Jul '19

10 Jul '19

Hello Ali, (please keep the sr-users list in CC, that other people can also participate) Remove the “-D” from your Kamailio command line – then the warning will disappear. This is also necessary because you want to have Kamailio listen on several addresses. If you want to test it more in a production like setting, you should also remove the “-E” from the command line. You will find the logging in your system log then. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services From: ali mahfouz <arm037(a)usal.edu.lb> Sent: Wednesday, July 10, 2019 11:54 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] kamailio warning Dear Mr.Henning, I am working on high availability on two Kamailio servers connected to the database using keepalived and when I am testing Kamailio I got this issue and I have no idea about fixing it i am trying to see by this command kamailio -E -D -f /etc/kamailio/kamailio.cfg 0(16101) INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: 10.10.10.20:5060<http://10.10.10.20:5060> udp: 10.10.10.3:5060<http://10.10.10.3:5060> tcp: 10.10.10.20:5060<http://10.10.10.20:5060> tcp: 10.10.10.3:5060<http://10.10.10.3:5060> Aliases: tcp: 10.10.10.3:5060<http://10.10.10.3:5060> tcp: 10.10.10.20:5060<http://10.10.10.20:5060> udp: 10.10.10.3:5060<http://10.10.10.3:5060> udp: 10.10.10.20:5060<http://10.10.10.20:5060> *: sip.domain.local:* WARNING: no fork mode and more than one listen address found (will use only the first one) 0(16101) INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(16101) INFO: jsonrpcs [jsonrpcs_sock.c:197]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... 0(16101) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(16101) INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available 0(16101) INFO: <core> [main.c:2669]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(16101) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(16101) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(16101) WARNING: <core> [main.c:1347]: main_loop(): using only the first listen address (no fork) 6(16131) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 5(16130) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/16130 ^C 6(16131) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 5(16130) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 1(16126) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16131 exited normally, status= 3(16128) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16126 exited normally, status= 4(16129) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16128 exited normally, status=0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16129 exited normally, status=0 0(16101) ALERT: <core> [main.c:751]: handle_sigs(): child process 16130 exited normally, status=0 0(16101) INFO: <core> [main.c:778]: handle_sigs(): dont_fork turned on, living on 0(16101) INFO: <core> [main.c:772]: handle_sigs(): SIGCHLD received, but no child has stopped, ignoring it 0(16101) NOTICE: <core> [main.c:709]: handle_sigs(): Thank you for flying kamailio!!! 2(16127) INFO: <core> [main.c:836]: sig_usr(): signal 15 received 0(16101) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Please help thank you for giving time from your time On Wed, Jul 10, 2019 at 12:40 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Ali, the first warning is related about the starting mode of Kamailio. You should not start it with “no forking mode” if you want to get rid of it. Have a look to your command line parameters: “-D” is the one to change. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of ali mahfouz Sent: Wednesday, July 10, 2019 8:46 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>; sr-users-request(a)lists.kamailio.org<mailto:sr-users-request@lists.kamailio.org>; sr-users-owner(a)lists.kamailio.org<mailto:sr-users-owner@lists.kamailio.org> Subject: [SR-Users] kamailio warning WARNING: no fork mode and more than one listen address found (will use only the first one) 0(8777) INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(8777) INFO: jsonrpcs [jsonrpcs_sock.c:197]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... 0(8777) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(8777) INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available 0(8777) INFO: <core> [main.c:2669]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(8777) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(8777) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(8777) WARNING: <core> [main.c:1347]: main_loop(): using only the first listen address (no fork) 5(8782) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/8782 6(8783) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) ^C 6(8783) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 5(8782) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 4(8781) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 3(8780) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 1(8778) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8782 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8778 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8781 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8783 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8780 exited normally, status=0 0(8777) INFO: <core> [main.c:778]: handle_sigs(): dont_fork turned on, living on 0(8777) INFO: <core> [main.c:772]: handle_sigs(): SIGCHLD received, but no child has stopped, ignoring it 0(8777) NOTICE: <core> [main.c:709]: handle_sigs(): Thank you for flying kamailio!!! 2(8779) INFO: <core> [main.c:836]: sig_usr(): signal 15 received 0(8777) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

3 3

(no subject)
by Ilie Soltanici 10 Jul '19

10 Jul '19

Hello, Could someone help me to set a variable as a string in Kamailio? At the moment i'm using the following statement: *$var(account)=$(au{s.substr,0,8}); * *xlogl("L_INFO", "[1234] Account: $var(* *account* *)\n");* which is working fine, but i need to change it into: *$var(account)=$(au{s.select,0,-}{s.toupper})-$(au{s.select,1,-});* By doing this I'm getting the error: *[core/rvalue.c:1007]: rval_get_int(): automatic string to int conversion for "AT" failed[core/rvalue.c:1912]: rval_expr_eval_int(): rval expression conversion to int failed (729,30-729,59)* I tried also: *$var(account)="" + "$(au{s.select,0,-}{s.toupper})-$(au{s.select,1,-})";* But, then I'm getting: *Account = $(au{s.select,0,-}{s.toupper})-$(au{s.select,1,-})* instead of the value. What is the correct way to set it? Thank you.

2 3

kamailio warning
by ali mahfouz 10 Jul '19

10 Jul '19

WARNING: no fork mode and more than one listen address found (will use only the first one) 0(8777) INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(8777) INFO: jsonrpcs [jsonrpcs_sock.c:197]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... 0(8777) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(8777) INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available 0(8777) INFO: <core> [main.c:2669]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(8777) INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(8777) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(8777) WARNING: <core> [main.c:1347]: main_loop(): using only the first listen address (no fork) 5(8782) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/8782 6(8783) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) ^C 6(8783) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 5(8782) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 4(8781) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 3(8780) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 1(8778) INFO: <core> [main.c:836]: sig_usr(): signal 2 received 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8782 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8778 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8781 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8783 exited normally, status=0 0(8777) ALERT: <core> [main.c:751]: handle_sigs(): child process 8780 exited normally, status=0 0(8777) INFO: <core> [main.c:778]: handle_sigs(): dont_fork turned on, living on 0(8777) INFO: <core> [main.c:772]: handle_sigs(): SIGCHLD received, but no child has stopped, ignoring it 0(8777) NOTICE: <core> [main.c:709]: handle_sigs(): Thank you for flying kamailio!!! 2(8779) INFO: <core> [main.c:836]: sig_usr(): signal 15 received 0(8777) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

2 1

mi_datagram module in ipv6
by Albert Petit 09 Jul '19

09 Jul '19

hi i would need to replicate behaviour of refresh watchers command at mi_datagram module in an ipv6 system. but in such system i am not able to bind this module in an iov6 address desoite its doc claims is possible https://kamailio.org/docs/modules/4.4.x/modules/presence.html#idp21723212 hence not sure i will be able to use mi_datagram using ipv6 address unless some user tells me he/shr could do succesfully as module is quite old question is: h¥w could i force kamailio to notify watchers of a subscriber in some other way from an external application? any hints? thanks

2 2

kamctl ul command seems NOK when using an IPv6 Address
by Albert Petit 09 Jul '19

09 Jul '19

Hi, I am trying to create an identity with permanent location in my system using kamctl ul add command. Usually i have no issue doing that in IPv4 ,example kamctl ul add mcpttcontrolling sip:mcpttcontrolling@127.0.0.1:5070 it is OK However when i do that using an IPv6 address i get following error: kamctl ul add mcpttcontrolling sip:mcpttcontrolling@ [2001:470:1f21:219:20c:29ff:fedb:db6e]:5070 error: invalid SIP URI: sip:mcpttcontrolling@ [2001:470:1f21:219:20c:29ff:fedb:db6e]:5070 Let me know what you would need to investigate or what i am doing wrong for adding a location with IPv6 Address using kamctl tool Thanks Albert

2 2

Jansson_get issues
by Igor Olhovskiy 09 Jul '19

09 Jul '19

Hi! I'm trying to get jannson_get working, but seems missing something. Example is quite simple, actually taken from wiki $var(result) = $null; jansson_get("pi", '{"mystr":"my input string", "count":9000, "mybool":true, "pi":3.14159}', $var(result)); When running this part of code getting ERROR: <core> [core/action.c:1139]: do_action(): runtime fixup failed for jansson_get param 3 ERROR: <core> [core/action.c:1518]: do_action(): run action error at: /etc/kamailio/kamailio.cfg:1030 Also tried with jannson_get(".pi", ...); jannson_get('/"pi"', ...); and a bit others. What am I missing? kamailio 5.2.3 Thanks!

2 3

KW 2019 presentation files
by Denys Pozniak 09 Jul '19

09 Jul '19

Hello! Where can I find fresh presentation files from the KW 2019? PS. Nothing here: https://www.kamailio.org/events/ -- BR, Denys Pozniak

3 3

where can i find siremis user guide?
by quxiao＠unioncast.cn 09 Jul '19

09 Jul '19

hi�� i have installed siremis successfully for the first time, but there is no guide to use siremis, where can i find it, thanks Best Regards & Thanks�� ֻ��13401129519 ��䣺quxiao(a)unioncast.cn ְλ��߼�C++��ʦ ��ţ��з�� Vҕͨ�_��־W�j�Ƽ��޹�˾ Unioncast Digital Network Technology Co., Ltd. Beijing ��˾��ҳ��http://www.unioncast.cn/ һ��ƶ��Ӧ�÷��ṩ�� ַ��к��йش��ϴ��3�ź��Ƽ��9��910�� ʱࣺ100081

1 0

Kamailio new in devel for upcoming 5.3.0
by Henning Westerholt 08 Jul '19

08 Jul '19

Hello, I have updated the "new in devel" features document in our wiki. Pretty amazing stuff already in for the upcoming 5.3.0: * 5 new modules * Changes in 32 existing modules * Many changes in core, API refactoring and tools * About 1000 commits from 62 authors (development still ongoing!) Full list at: https://www.kamailio.org/wiki/features/new-in-devel A summary can be found in my today talk at COMMCON, slides 10,11: https://skalatan.de/de/archiv/presentations/commcon-2019-presentation.pdf Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 1

How to statelessly send a reply to custom destination?
by George Diamantopoulos 05 Jul '19

05 Jul '19

Hello all, I've been trying to handle OPTIONS statelessly, but proper handling fails for UACs behind NAT without ALG. I thought I'd overcome this by setting the $du pseudovariable, but apparently I'm missing something. The following does not work: if ($Rp == "6050") { force_send_socket(udp:10.10.10.10:6050); } $du = "sip:" + $si + ":" + $sp; sl_send_reply("200","OK"); exit; I'm now realising that I put this in request_route() so force_send_socket() and $du have no use whatsoever, since the request is not forwarded anywhere. Is this the reason? So the question is, how can I do the following things for a stateless reply: a) choose a sending socket for the reply and b) change the destination for the reply to something other than the URI in the first Via header. Thanks! BR, George

3 3

Append a new branch while the other one is in ringing state
by Koray Vatansever 04 Jul '19

04 Jul '19

Hi all, I want to fork a new branch, while the first one is ringing. Forking will take place in a rtimer process. So I will save the invite transaction in a htable, and using this transaction info I need to append a branch. I saw TSILO module provides some methods to append branch. However documentation says that it can be used when register message received. What is the best way to implement this? Thanks, Koray

3 5

people complain Kamailio not handling stale nonce correctly
by Juha Heinanen 03 Jul '19

03 Jul '19

Several people are complaining on baresip git that Kamailio is among those SIP servers that don't handle stale nonce properly, i.e., don't include stale=true flag in WWW-Authenticate header of 401 response. I have tried to tell that Kamailio does what it has been configured to do, but the response is that "by default" Kamailio works incorrectly. "By default" I guess refers to the example Kamailio kamailio.cfg in etc dir. How about adding stale nonce handling to the example config, for example, like in below? -- Juha *** kamailio.cfg 2019-06-13 17:17:17.182148989 +0300 --- /tmp/kamailio.cfg 2019-07-02 18:02:02.947183503 +0300 *************** *** 745,753 **** if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ! if (!auth_check("$fd", "subscriber", "1")) { ! auth_challenge("$fd", "0"); ! exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) --- 745,759 ---- if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ! switch (auth_check("$fd", "subscriber", "1")) { ! case 1: ! break; ! case -4: ! auth_challenge("$fd", 17); ! exit; ! default: ! auth_challenge("$fd", "0"); ! exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH"))

3 12

Change
by Samir Kundu 03 Jul '19

03 Jul '19

Please note change of my new email address – samirkundu(a)samirkundultd.plus.com Thank you Samir ====================================================================== COMPUTING & COMMUNICATION HARDWARE & SOFTWARE SOLUTIONS PROVIDER ====================================================================== e-mail is confidential --- "SAVE PAPER - THINK BEFORE YOU PRINT!" Samir Kundu Ltd | Tel: 0191 538 8280 | Email: samir_kundu(a)samirkundugroup.co.uk Address: 3 Beechwood Terrace, Sunderland SR2 7LY Registered in England & Wales No. 09051111 ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

2 1

DMQ: dmq_t_replicate()
by Lars Olsson 02 Jul '19

02 Jul '19

Hi, I am trying to replicate a REGISTER request between several nodes. When I am using dmq_t_replicate(), it ends the execution of the script. It this correct or I have missed anything? I can not find anything in the documentation of the DMQ module the says that it will end the script execution. Is it recommended to use the DMQ module or should I simply forward the request instead as described in https://medium.com/@tumalevich/kamailio-registration-replication-without-dm… Best Regards, Lars

2 3

[DMQ] Sync DLG_STATE_CONFIRMED_NA dialog state ?
by Mathieu Bodjikian 02 Jul '19

02 Jul '19

Hello, We are using DMQ module to sync dialogs between 3 Kamailio boxes. We are using last Kamailio version (5.2.3). >From our understanding, the state DLG_STATE_CONFIRMED_NA is not synced over DMQ (https://github.com/kamailio/kamailio/blob/a84a3ea618f0e602a8892c37fce7f4e72…) With containerized environement, the ACK's can go to any Kamailio instance, which lead to : On first box : -> INVITE received -> Dialog created + KDMQ sent (state DLG_STATE_UNCONFIRMED) -> 180 | 183 received -> Dialog updated + KDMQ sent (state DLG_STATE_EARLY) -> 200 received -> Dialog updated + KDMQ NOT sent (DLG_STATE_CONFIRMED_NA) On second box : -> ACK received : - next_state_dlg(): bogus event 6 in state 2 for dlg - dialog not updated - KDMQ with state DLG_STATE_CONFIRMED not sent On first box, after timeout : -> After timeout : - dialg with incorrect timeouts (since box2 didn't send dialog update) - tm sends BYE in both ways - bye_reply_cb(): inconsitent dlg timer data on dlg We see two solutions : - We add DLG_STATE_CONFIRMED_NA to states we sync on dlg_dmq_replicate_action - We manage to route the ACK on the same instance that received first INVITE (could be tricky) Which one is preferable ? Kind regards, Mathieu Bodjikian

3 2

question about reply route
by Iván Aponte 02 Jul '19

02 Jul '19

Hello, I have a question about record route and via headers. My scenario is the following: [asterisk] ---- [kam] -------- [trunk] 10.142.0.27 10.142.0.6 200.x.x.x I have an asterisk an a kamailio in a private ip network (10.142.0.0/24). Kamailio has the address 10.142.0.6 but is also natted 1:1 to a public ip (35.0.0.6). My problem is that when the invite are replied the asterisk replies to the external ip (35.0.0.6) instead of the internal ip (even though the kamailio replies goes through the internal network) . I tried adding a reply route to change the advertised address to no avail. any advice would be appreciated. this is my conf: LISTEN="listen=udp:0.0.0.0:5060 advertise 35.0.0.6:5060" disable_tcp=yes loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "permissions.so" loadmodule "ipops.so" modparam("jsonrpcs", "pretty_format", 1) modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 0) reply_route { xinfo("source $si"); if(!is_in_subnet($si, "10.0.0.0/8")) { set_advertised_address("10.142.0.6"); } else { xinfo("not in subnet"); } } request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # 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(); } # dispatch requests to foreign domains route(SIPOUT); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } } # Wrapper for relaying requests route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); 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()) { if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { send_reply("405", "Method Not Allowed"); 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; } route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } -- Iván Aponte Office: +58(212)9923193 Mobile: +58(412)2774713

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Re: [SR-Users] sr-users Digest, Vol 170, Issue 1
by ali mahfouz 02 Jul '19

02 Jul '19

chat over sip where and how save it On Mon, Jul 1, 2019 at 1:03 PM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: ATCF/ATGW functionality (Mojtaba) > 2. Re: ATCF/ATGW functionality (Mojtaba) > 3. Re: ATCF/ATGW functionality (Maryam Baghdadi) > 4. kamailio wiht PJSIP IOS client (Gaurav Bmotra) > 5. Re: ATCF/ATGW functionality (Mojtaba) > 6. Re: Late offer ACKs not passing through branch_route > (Enrico Bandiera) > 7. kamailio 4.4.4 fills IPv6 contact address when sending packet > over IPv4 (Pali Rohár) > 8. Re: install mongoDB module with kamailio-cli doker image > (Gaurav Bmotra) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 30 Jun 2019 15:40:07 +0430 > From: Mojtaba <mespio(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Cc: users(a)lists.kamailio.org > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > < > CABVi_Ew5Ti0bH5z4kcoBwx9jvv8S06GtEqgzCKPfxE7jfh7aJQ(a)mail.gmail.com> > Content-Type: text/plain; charset="UTF-8" > > Unfortunately you can't, because SRVCC is a capable between MME and > MCS in your network. When the UE is moving on, the MME send "Take > handover of the call" to MSC and move the surrect session control to > MSC. You could read more about SRVCC in 3GPP TS 29.280. > And FHOSS does not support interface between MME and HSS, > With Best Regards.Mojtaba > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > <maryam.baghdadi(a)gmail.com> wrote: > > > > Hello, > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > --Mojtaba Esfandiari.S > > > > ------------------------------ > > Message: 2 > Date: Sun, 30 Jun 2019 15:40:07 +0430 > From: Mojtaba <mespio(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Cc: users(a)lists.kamailio.org > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > < > CABVi_Ew5Ti0bH5z4kcoBwx9jvv8S06GtEqgzCKPfxE7jfh7aJQ(a)mail.gmail.com> > Content-Type: text/plain; charset="UTF-8" > > Unfortunately you can't, because SRVCC is a capable between MME and > MCS in your network. When the UE is moving on, the MME send "Take > handover of the call" to MSC and move the surrect session control to > MSC. You could read more about SRVCC in 3GPP TS 29.280. > And FHOSS does not support interface between MME and HSS, > With Best Regards.Mojtaba > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > <maryam.baghdadi(a)gmail.com> wrote: > > > > Hello, > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > --Mojtaba Esfandiari.S > > > > ------------------------------ > > Message: 3 > Date: Sun, 30 Jun 2019 16:32:01 +0430 > From: Maryam Baghdadi <maryam.baghdadi(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] ATCF/ATGW functionality > Message-ID: > <CAPrdAbAnnTULTA1rZ-b= > PG1FE+4HfqRHtKg_3R49WEBUXbM+Fw(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Thank you for your response, > If we need to handover of call in our network, What's the solution? is it > possible to add external MSC Server to out network? or add a SBC with this > ability ? > > On Sun, Jun 30, 2019 at 3:40 PM Mojtaba <mespio(a)gmail.com> wrote: > > > Unfortunately you can't, because SRVCC is a capable between MME and > > MCS in your network. When the UE is moving on, the MME send "Take > > handover of the call" to MSC and move the surrect session control to > > MSC. You could read more about SRVCC in 3GPP TS 29.280. > > And FHOSS does not support interface between MME and HSS, > > With Best Regards.Mojtaba > > > > On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi > > <maryam.baghdadi(a)gmail.com> wrote: > > > > > > Hello, > > > > > > Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? > > > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . > > > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks. > > > > > > _______________________________________________ > > > Kamailio (SER) - Users Mailing List > > > sr-users(a)lists.kamailio.org > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > --Mojtaba Esfandiari.S > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > >

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kamailio wiht PJSIP IOS client
by Gaurav Bmotra 01 Jul '19

01 Jul '19

hi i na using kamailio 5.1.2 and i m developing ios client with PJSIP audio is working fine ..but on video it work one side but not showing other side it is working fine on other client like zoiper for audio/video both can anyone help me -- *Regards:* Gaurav Kumar

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IMS strange behaviour
by Amar Tinawi 01 Jul '19

01 Jul '19

Hello list i'm facing strange behavior in my lap which consists of p-cscf, i-cscf, s-cscsf, FHoSS for HSS Registration is ok and ICSCF is performing well but when it comes to calls, INVITE Call between two ims users is not passing from scscf to icscf the flow should be : P-->S-->I-->hss i can see invite massage is going out scscf to icscf with all monitoring tools, but no incoming Invite massage to icscf not on sip level and nor on networking level (tcpdump) Actually i don't know where the INVITE packets are going Any advice Regards

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install mongoDB module with kamailio-cli doker image
by Gaurav Bmotra 01 Jul '19

01 Jul '19

hi i wan to install mongodb module in kamailio/kamailio-ci image pl help me thnaks -- *Regards:* Gaurav Kumar

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Late offer ACKs not passing through branch_route
by Bandiera, Enrico 01 Jul '19

01 Jul '19

Hi, I'm encountering an issue when relaying late offer ACKs, before relaying I set the branch_route with t_on_branch() but kamailio never enters that route after calling t_relay I'm keeping the rtpengine module calls inside branch_routes to avoid issues when re-routing in failure routes I suppose it's something related to ACK packets but I'm not sure what the issue is. Any hints? Thanks, Enrico.

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ATCF/ATGW functionality
by Maryam Baghdadi 01 Jul '19

01 Jul '19

Hello, Is it possible to add ATCF/ATGW functionality in kamailio as P-cscf ? I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss . ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks.

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Kamailio 5.x and Call_control module
by Efelin Novak 30 Jun '19

30 Jun '19

Hi Folks, I'm migrating to Kamailio 5.2.x from 4.4.x. Everything seems to be fine, however I have came to an issue with call_control module as this one is still using old MI interface. Standard situations work nice (maximum debit, prepaid, "CDRs") however when call_control needs to kill a call (credit is gone), it tries to send dlg_end_dlg over MI and it fails. Question are: Is call_control still supported? I haven't found any note about it. Is there any workaround from Kamailio point of view? I'm running last versions of both applications. I'm also rising a ticket at ag-projects side. Thanks Efelin

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Re: [SR-Users] sr-users Digest, Vol 169, Issue 27
by ali mahfouz 29 Jun '19

29 Jun '19

if I send my kamailio.cfg file can you edit it to me please because it is so urgent On Thu, Jun 27, 2019 at 1:06 PM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: TOPOS uid in contact param instead of username > (Daniel-Constantin Mierla) > 2. Kamailio 5.x and Call_control module (Efelin Novak) > 3. Re: Kamailio 5.x and Call_control module > (Daniel-Constantin Mierla) > 4. Statistics for locally generated replies (Duarte Rocha) > 5. Re: Kamailio 5.x and Call_control module > (Daniel-Constantin Mierla) > 6. Re: Kamailio 5.x and Call_control module (Efelin Novak) > 7. Re: kamailio as UAC and NAPTR / sendsocket question > (Karsten Horsmann) > 8. MongoDB for Kamailio (Gaurav Bmotra) > 9. Presence publish manipulation (Gertjan Wolzak) > 10. Re: Question about registrar behavior > (Володимир Іванець) > 11. Re: MongoDB for Kamailio (Henning Westerholt) > 12. Re: DMQ: dmq_t_replicate() (Henning Westerholt) > 13. Re: does kamailio support graph database ?? (Henning Westerholt) > 14. Re: Question about registrar behavior (Henning Westerholt) > 15. Need help with "no branches for forwarding" message (Andrew Chen) > 16. Topos / Remove HF_re (Nicolas Breuer) > 17. Re: Kamailio 5.x and Call_control module (Efelin Novak) > 18. Re: Question about registrar behavior > (Володимир Іванець) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Jun 2019 12:52:05 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] TOPOS uid in contact param instead of username > Message-ID: > < > CAFRry4VMJd9aaJ_teWd3E-CYg-O9K+qutW699n9GDc9is7wAxQ(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hello, > ok, feel free to ask question if something is not clear ... and I think > topoh might help looking at its code, iirc, it uses uri param for contact > masking ... > > Anyhow, there is no better fun that C coding ;-) ... > Cheers, > Daniel > > On Wed, Jun 26, 2019 at 10:12 AM Thomas Weber <thomas.weber(a)pascom.net> > wrote: > > > Hello, > > > > thank you for your opinions. > > > > So i will give it a try throughout the next weeks because we have some > > customer pressure here. > > Didn't do a C project since some 10 years, so it could be fun to "learn > it > > again". > > > > Cheers, > > > > > > Thomas > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >

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Re: [SR-Users] Kamailio next_state_dlg(): bogus event
by Olli Attila 29 Jun '19

29 Jun '19

Hello, The prack is for the initial INVITE establishing the call. No re-INVITE messages are generated in the dialog. --Olli Message: 11 Date: Fri, 28 Jun 2019 08:35:09 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamailio next_state_dlg(): bogus event Message-ID: <CAFRry4XAF5qYqRxgJA4tzsuUt-56ON1KnzzwvJnxeKOdTUaxSA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hello, is the PRACK following a re-INVITE or is for the initial INVITE establishing the call? Cheers, Daniel On Fri, Jun 28, 2019 at 8:13 AM Olli Attila <attiolli(a)gmail.com> wrote: > Hello, > > I'm getting this error in Kamailio log every now and then. Can anyone > suggest methods to debug this further for root cause? > > Jun 27 10:04:09 /usr/sbin/kamailio[26341]: CRITICAL: dialog > [dlg_hash.c:1098]: next_state_dlg(): bogus event 5 in state 4 for dlg > 0x7fceeabf8d38 [749:3185] with clid > 'SDceked01-e4d2e93d7f9ffe88907206a25ccdbcc2-v300g00030' and tags > 'SDceked01-ecd4281b' '14001358846426' > > Behaviour in the call: > When I look at the SIP flow from this call I can see that a PRACK > message which is routed to Kamailio gets discarded (=the proxy does > not route the PRACK onwards to the UA that requested the usage of the > 100rel in 183 session progress message.) The PRACK message itself > seems valid and when compared to a PRACK from a working call, there is > no difference in the content. > > Cheers, > Olli

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Htable not reloading from db on expiry
by Daniel Greenwald 28 Jun '19

28 Jun '19

I have an htable that loads from mysql. The data loads fine on startup but disappears on expiry. If I issue kamcmd htable.reload it loads again until expiry. It seems as if db_expires is not taking affect. I see them expiring in the debug log but not an errors. Mysql general log does not show any attempt to connect on expiry. Another odd thing is that if I set the expires column in db to anything other than 0 the data doesn't load at all. Any ideas would be appreciated. modparam("htable", "timer_interval", 10) modparam("htable", "htable", "h=>size=20;autoexpire=20;dbtable=validdomains;updateexpire=0;") modparam("htable", "db_url", AUTHDBURL) modparam("htable", "db_expires", 1) create or replace view validdomains as select distinct subscriber.domain as key_name, 1 as key_value, 0 as key_type, 0 as value_type, 0 as expires from subscriber;

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Kamailio next_state_dlg(): bogus event
by Olli Attila 28 Jun '19

28 Jun '19

Hello, I'm getting this error in Kamailio log every now and then. Can anyone suggest methods to debug this further for root cause? Jun 27 10:04:09 /usr/sbin/kamailio[26341]: CRITICAL: dialog [dlg_hash.c:1098]: next_state_dlg(): bogus event 5 in state 4 for dlg 0x7fceeabf8d38 [749:3185] with clid 'SDceked01-e4d2e93d7f9ffe88907206a25ccdbcc2-v300g00030' and tags 'SDceked01-ecd4281b' '14001358846426' Behaviour in the call: When I look at the SIP flow from this call I can see that a PRACK message which is routed to Kamailio gets discarded (=the proxy does not route the PRACK onwards to the UA that requested the usage of the 100rel in 183 session progress message.) The PRACK message itself seems valid and when compared to a PRACK from a working call, there is no difference in the content. Cheers, Olli

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Statistics for locally generated replies
by Duarte Rocha 27 Jun '19

27 Jun '19

Greetings, Does the statistics command "kamcmd tm.stats" has an argument or a way to only list locally generated replies? As far as i know, sl.stats only lists local generated replies. Can tm work the same way? Best Regards, Duarte Rocha

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Need help with "no branches for forwarding" message
by Andrew Chen 27 Jun '19

27 Jun '19

Hey guys, I need help to identify the cause of this is error: Jun 26 12:03:53 ashintgtpsg51 /usr/sbin/kamailio[1651]: ERROR: tm [t_fwd.c:1728]: t_forward_nonack(): no branches for forwarding This message was shortly printed after I xlog a message in onsend_route[] block. What else is strange is that the dialog continued on with that same SIP Call-ID. This is what it looks like in the wireshark flow diagram: [image: image.png] The dispatcher entry is an SRV record which lists available hosts. Any thoughts on how I can identify the cause of this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) achen@ <achen(a)thinkingphones.com>fuze.com -- *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.*

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Topos / Remove HF_re
by Nicolas Breuer 27 Jun '19

27 Jun '19

Hello, When using TOPOS module, Kamailio add this header : P-SR-XBranch before executing the topos module. Doing this "remove_hf_re("P-")" in the scripting seems to bypass the addition of the P-SR-XBranch in the sip message. In my mind, remove header function just remove the headers based on the incoming message. What do you think ?

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Question about registrar behavior
by Lars Olsson 27 Jun '19

27 Jun '19

Hi, I have found a behavior in the registrar module that I do have a question about. Is the current behavior correct and wanted? Using the save() method in the script I see the following: * Processing a register request for a user gives return code 1 ( or 2 ) * Processing a unregister request (expires=0) for registered user gives return code 3 * Processing a unregister request (expires=0) for a user which is NOT registered gives return code 1. Why? What is the reason behind this? No database entry is added which is expected. Test performed on 5.1.4, using DB mode 3. For handling a late unregister request ( where registration has already expired) return code does not reflect the the action. I assume that manually checking $expires(max) is the option to go then if I want to detect the unregister request or? Cheers, Lars

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does kamailio support graph database ??
by Gaurav Bmotra 26 Jun '19

26 Jun '19

hi i m using kamailio 5.1 , does kamailio support graph database ?? thank you -- *Regards:* Gaurav Kumar

2 1

MongoDB for Kamailio
by Gaurav Bmotra 26 Jun '19

26 Jun '19

hi i m using kamailio 5.1 , on Ubuntu 18.4 and right now i m using Maria DB for kamailio , but i want to migrate my Maria data to MingoDB ,, plz help me how can i use MOngoDB with kamailio thanks -- *Regards:* Gaurav Kumar

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Presence publish manipulation
by Gertjan Wolzak 26 Jun '19

26 Jun '19

Hello Kamailions, I'm running into the following challenge. A kamailio setup where phones register with their mac address as uid and a password. Users have to login to the system by dialing a special number where they have to enter their internal extension and a pin code. Now the system knows which extension(number) is logged in on which phone(macaddress). This is done with from updates, just a type of hotdesking. Now I want to use presence.... When a phone makes a phone call, kamailio generates a publish message where the "local" part of the xml code contains the mac address. How can I manipulate the Publish xml information? I mean the manipulation can be done with textops, but which variables to use and where in the configuration? Im asking because when I disable the presence route and do not call it in my configuration, but with presence enabled, I still get Publish messages.... I have tried adding a Sender header which contains the "extension" of the phone and then call the publish in the PRESENCE route like this: handle_publish("$hdr(Sender)"); But this does not change the xml information. Am I doing things wrong? Below the first publish message that is send, where 10.88.77.172 is the kamailio server. U 2019/06/26 14:54:37.233345 10.88.77.172:5060 -> 10.88.77.172:5060 #7 PUBLISH sip:0015659a9931@10.88.77.172:5060 SIP/2.0. Via: SIP/2.0/UDP 10.88.77.172;branch=z9hG4bK1a87.5510d093000000000000000000000000.0. To: <sip:0015659a9931@10.88.77.172:5060>. From: <sip:0015659a9931@10.88.77.172:5060>;tag=a2652bb825a097f3b7285d4c70edd51a-5c5d. CSeq: 10 PUBLISH. Call-ID: 173009c21f4e3c77-94480(a)10.88.77.172. Content-Length: 579. User-Agent: kamailio (5.1.8 (x86_64/linux)). Max-Forwards: 70. Event: dialog. Expires: 125. Content-Type: application/dialog-info+xml. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:0015659a9931@10.88.77.172:5060"> <dialog id="0_2285997339(a)10.88.77.184" call-id="0_2285997339(a)10.88.77.184" direction="initiator"> <state>Trying</state> <remote> <identity>sip:604113@10.88.77.172:5060</identity> <target uri="sip:604113@10.88.77.172:5060"/> </remote> <local> <identity>sip:0015659a9931@10.88.77.172:5060</identity> <target uri="sip:0015659a9931@10.88.77.172:5060"/> </local> </dialog> </dialog-info> I would like to change the local identity and target uri into the extension assigned to that phone, for example 604114.... So that subscribers to the extension 604114 get informed about the state of the phone with the used uid 0015659a9931. Feedback would be appricated. Pointers to where I am going wrong as well, ... that's feedback too. Rgds, Gertjan

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TOPOS uid in contact param instead of username
by Thomas Weber 26 Jun '19

26 Jun '19

Hello, we're using topos/redis since some time and it's working quite well for most situations. It's used to route asterisk traffic to sip providers. Some providers, however, are validating the contact header strictly against i.e. the accounts username and so we have to disable topos by using event_route[topos:msg-sending]. Now i'm wondering if it would be useful to other people to add a mode to topos where the topos uid is not forwarded as contact username but instead added as a contact uri param. So instead of e.g. Contact: <sip:btpsh-5abcdef5-374-1@x.x.x.x:5060;transport=tcp> it would use Contact: <sip:+49......@x.x.x.x:5060;transport=tcp;line=btpsh-5abcdef5-374-1>. What do you think? Would it make sense? Is it a lot of effort? Cheers, Thomas

4 4

Apache version for FHoSS
by Woscek, Martin W. 26 Jun '19

26 Jun '19

Hello all, What is the version of Apache that FHoSS was built to? Thanks, Martin

2 1

Info: ClueCon Weekly - Kamailio Project Updates
by Daniel-Constantin Mierla 25 Jun '19

25 Jun '19

Hello, later today I will join ClueCon Weekly for a discussion about what is new for Kamailio project and the plans for the future. ClueCon Weekly is a web video conference organized by the FreeSwitch project team. They announced recently a revamp of the format, however, I expect that other people can still join live if they want to listen and ask questions, couple of ways to connect being listed at: * https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… If joining live is no longer possible, I will post a link to the recording and if someone has any questions following my presentation, I can answer here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 4

Carrier Route installation Failed
by Amar Tinawi 24 Jun '19

24 Jun '19

Hello i'm facing a dependency error when trying to install kamailio 5.2.3+xenial with carrier route module using apt package installer The following packages have unmet dependencies: kamailio-carrierroute-modules : Depends: kamailio (= 4.3.4-1.1ubuntu2) but 5.2.3+xenial is to be installed E: Unable to correct problems, you have held broken packages. all other modules installed fine any suggestions? thanks in advance

4 5

Kamailio routing BYE back to originator
by Andrew White 24 Jun '19

24 Jun '19

Hi team! I’ve recently found an issue in my Kamailio setup. It appears when the trunk replies to call in progress with a BYE, we relay it immediately back to them: https://i.imgur.com/h5fusau.png <https://i.imgur.com/h5fusau.png> The only differences in the second BYE is the Route header is removed and replaced with a Via header referencing the Kamailio machine, and the From URI is rewritten to show the Kamailio hostname. I’m unsure what steps I can take here, as all of my out of dialog messages (180, 200, ACK) are forwarded along with no issue. It seems only the in dialog are having this issue. Here’s my WITHINDLG and RELAY: def ksr_route_relay() if KSR.is_method_in("IBSU") then if KSR::TM.t_is_set("branch_route") < 0 then KSR::TM.t_on_branch("ksr_branch_manage") end end if KSR.is_method_in("ISU") then if KSR::TM.t_is_set("onreply_route") < 0 then KSR::TM.t_on_reply("ksr_onreply_manage") end #KSR.info("Onreply route: #{KSR::TM.t_is_set("onreply_route")}") end if KSR.is_INVITE() then if KSR::TM.t_is_set("failure_route") < 0 then KSR::TM.t_on_failure("ksr_failure_manage") end end if KSR::TM.t_relay() < 0 then KSR::SL.sl_reply_error() end exit end def ksr_route_withindlg() return if KSR::SIPUTILS.has_totag() < 0 if KSR::RR.loose_route() > 0 then ksr_route_dlguri() if KSR.is_BYE() then #KSR.setflag($myenv['FLT_ACC'].to_i) #KSR.setflag($myenv['FLT_ACCFAILED'].to_i) elsif KSR.is_ACK() then ksr_route_natmanage() elsif KSR.is_NOTIFY() then KSR::RR.record_route() end ksr_route_relay() exit end if KSR.is_ACK() then #KSR.info("Handling an ACK within dialog") if KSR::TM.t_check_trans() > 0 then ksr_route_relay() exit else exit end end #KSR.info("404ing within dialog") KSR::SL.sl_send_reply(404, "Not here"); exit end I suspect I must be sending this to the wrong branch somehow, but I’m unsure how to correct this. The call gets originally initiated from Kamailio writing a custom RURI and To header before calling t_relay(). I’m wondering if I’m somehow initiating the dialog incorrectly and as such the BYE doesn’t know where to go. Thanks! Andrew

2 3

Kamailio config file flow chart
by Gaurav Bmotra 22 Jun '19

22 Jun '19

hi i m looking for kamailio config file flow chart ,,, if any one have plz provide thanks -- *Regards:* Gaurav Kumar

2 1

5.2.3 - dispatcher module looses setid
by Andrew Chen 21 Jun '19

21 Jun '19

Hi all, For a few times now I've noticed Kamailio 5.2.3 dispatcher module looses view of setid destination and this system been up and running for 13 days: root:~# kamcmd dispatcher.list | grep ash.tpgw.service To work around it, we have to reload the module again: root:~# kamcmd dispatcher.reload root:~# kamcmd dispatcher.list | grep ash.tpgw.service URI: sip:<some host>:5062 URI: sip:<some host>:5060 root:~# Process been running for several days: root:~# ps -elf | grep kam 0 S root 8484 6520 0 80 0 - 3714 pipe_w 16:04 pts/0 00:00:00 grep --color=auto kam 5 S kamailio 32086 1 0 80 0 - 214845 pause Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32164 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32165 32086 0 80 0 - 233342 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32166 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32167 32086 0 80 0 - 233341 skb_wa Jun17 ? 00:00:02 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32168 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32169 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio 1 S kamailio 32170 32086 0 80 0 - 214875 skb_wa Jun17 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.cfg -m 512 -M 64 -u kamailio -g kamailio Anyone seen this? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *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 3

Kamailio doesn't start with TLS
by Олег Подгуйко 21 Jun '19

21 Jun '19

Im trying to use kamailio with websocket and tls. It will be use as registrar for websocket clients. I'm using default config with additional websocket, https and tls modules. I created some certs for it and everething looks like good. But when I started kamailio via (systemctl start kamailio) I get some errors июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) DEBUG: <core> [core/tcp_main.c:2753]: tcp_init(): added 37.143.9.210:443 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR: <core> [core/tcp_main.c:2855]: tcp_init(): bind(9, 0x7fb6ef026f84, 16) on 37.143.9.210:443 : Permission denied июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) ERROR: tls [tls_init.c:342]: tls_h_init_si(): Error while initializing TCP part of TLS socket 37.143.9.210:443 июн 04 15:00:52 p534507.kvmvps kamailio[23502]: 0(23502) DEBUG: rtpengine [rtpengine.c:1986]: mod_destroy(): rtpengine_hash_table_destroy() success! If I start it via /usr/sbin/kamailio - all work. What I do wrong? -- Олег Подгуйко

5 8

kamcmd uac.* commands dont find records
by Karsten Horsmann 21 Jun '19

21 Jun '19

Good morning List! I have an issue with my kamcmd from kamailio 5.2.3 on CentOS 7.6 with kamailio-rpms. The kamcmd util can not find my registrations via command kamcmd uac.reg_enable auth_username FFFFFFFFFFFF kamcmd uac.reg_info l_username +49XXXXXXXX and soone. Only global commands for uac works. Whats going wrong with my kamcmd? Any ideas? Thank for your hints! # kamcmd uac.reg_dump { l_uuid: 49XXXXXXX l_username: +49XXXXXXXX l_domain: sip-trunk.telekom.de r_username: +49XXXXXXXX r_domain: sip-trunk.telekom.de realm: sip-trunk.telekom.de auth_username: FFFFFFFFFFFF auth_password: *************** auth_proxy: sip:reg.sip-trunk.telekom.de expires: 360 flags: 20 diff_expires: 129 timer_expires: 1561103682 reg_init: 1561101919 reg_delay: 3 } # kamcmd uac.reg_enable auth_username FFFFFFFFFFFF error: 404 - Record not found -- Kind regards Mit freundlichen Grüßen *Karsten Horsmann*

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Re: [SR-Users] FHoSS db web interface failing for retrievals -UPDATE
by Woscek, Martin W. 20 Jun '19

20 Jun '19

I can only compile down to JDK 1.6 java 6 but still get same error below from the web interface. But get the following error when pointing to JDK1.5 java 5 which would imply some parts of the code was built with 1.6. BTW I downgraded Ant to 1.9 (Apache Ant(TM) version 1.9.14 compiled on March 12 2019). Buildfile: /opt/OpenIMSCore/FHoSS/build.xml init: [mkdir] Created dir: /opt/OpenIMSCore/FHoSS/bin compile: [javac] /opt/OpenIMSCore/FHoSS/build.xml:48: warning: 'includeantruntime' was not set, defaulting to build.sysclasspath=last; set to false for repeatable builds [javac] Compiling 202 source files to /opt/OpenIMSCore/FHoSS/bin [javac] /opt/OpenIMSCore/FHoSS/src/de/fhg/fokus/hss/auth/AuthenticationVector.java:49: cannot access de.fhg.fokus.diameter.DiameterPeer.data.AVP [javac] bad class file: /opt/OpenIMSCore/FHoSS/lib/jdp.jar(de/fhg/fokus/diameter/DiameterPeer/data/AVP.class) [javac] class file has wrong version 51.0, should be 49.0 [javac] Please remove or make sure it appears in the correct subdirectory of the classpath. [javac] import de.fhg.fokus.diameter.DiameterPeer.data.AVP; [javac] ^ [javac] 1 error BUILD FAILED /opt/OpenIMSCore/FHoSS/build.xml:48: Compile failed; see the compiler error output for details. Any insight would be appreciated. Thanks, Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Woscek, Martin W. Sent: Tuesday, June 18, 2019 2:38 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] FHoSS db web interface failing for retrievals Hi all, We can enter user data into our HSS, we just cant retrieve data. Just to verify, FHoSS is based on Java 1.7 and java.lang.CharSequence is in Java 1.7? What we have: java version "1.7.0_80" Java(TM) SE Runtime Environment (build 1.7.0_80-b15) Java HotSpot(TM) 64-Bit Server VM (build 24.80-b11, mixed mode) The error: HTTP Status 500 - ________________________________ type Exception report message description The server encountered an internal error () that prevented it from fulfilling this request. exception org.apache.jasper.JasperException: Unable to compile class for JSP An error occurred at line: 116 in the jsp file: /pages/user/imsu_search_result.jsp Generated servlet error: The type java.lang.CharSequence cannot be resolved. It is indirectly referenced from required .class files org.apache.jasper.compiler.DefaultErrorHandler.javacError(DefaultErrorHandler.java:84) org.apache.jasper.compiler.ErrorDispatcher.javacError(ErrorDispatcher.java:328) org.apache.jasper.compiler.JDTCompiler.generateClass(JDTCompiler.java:397) org.apache.jasper.compiler.Compiler.compile(Compiler.java:288) org.apache.jasper.compiler.Compiler.compile(Compiler.java:267) org.apache.jasper.compiler.Compiler.compile(Compiler.java:255) org.apache.jasper.JspCompilationContext.compile(JspCompilationContext.java:556) org.apache.jasper.servlet.JspServletWrapper.service(JspServletWrapper.java:293) org.apache.jasper.servlet.JspServlet.serviceJspFile(JspServlet.java:291) org.apache.jasper.servlet.JspServlet.service(JspServlet.java:241) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) org.apache.struts.action.RequestProcessor.doForward(RequestProcessor.java:1063) org.apache.struts.tiles.TilesRequestProcessor.doForward(TilesRequestProcessor.java:263) org.apache.struts.action.RequestProcessor.processForwardConfig(RequestProcessor.java:386) org.apache.struts.tiles.TilesRequestProcessor.processForwardConfig(TilesRequestProcessor.java:318) org.apache.struts.action.RequestProcessor.process(RequestProcessor.java:229) org.apache.struts.action.ActionServlet.process(ActionServlet.java:1194) org.apache.struts.action.ActionServlet.doPost(ActionServlet.java:432) javax.servlet.http.HttpServlet.service(HttpServlet.java:709) javax.servlet.http.HttpServlet.service(HttpServlet.java:802) Thanks, _Martin

1 0

XMPP
by David Villasmil 19 Jun '19

19 Jun '19

Hello guys, I'm kind of confused as to what module exactly i need to user, let me explain: I have a service for DIDs and SMS from s.ms, i created a python script that connects to the service using xmpp to their jabber server with a user and a password, i.e.: xmpp_client.py MY-NUMBER(a)s.ms MY-PASSWORD What i do with that script is basically connect and wait to receive SMS, which i then insert in the proxy db (silo) and they get delivered by msilo. I have several numbers and doing this for every number seems stupid, so i started looking for a better solution, since the end user will be receive the message via SIP MESSAGE on they sip client. i came across kamailio/opensips' jabber and xmpp modules, but i'm not sure which (if any) to use, and even HOW to, or the actual architecture, since the documentation keeps talking about a local xmpp server? I would REALLY appreciate some help here. Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 2

Question on Database Version Table
by JR Richardson 18 Jun '19

18 Jun '19

Hey All, I'm doing some upgrades and noticed database table version numbers have increased. database table adjustments: | address | production ver 3 | upgrade 5.2 ver 6 | need to add accountcode and notes column | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add ruri_pattern and priority column I run mysql master/slave so I'm concerned about the master database not having the correct version numbers or can I just update the version number in the table when I do the upgrade of the kamailio versions? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

3 2

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