- sr-users - kamailio.org

set_advertised_address with variable
by Patrick Wakano 08 Oct '18

08 Oct '18

Hello list, Hope you all doing fine! I am using version 5.0.7 and I was trying to use the set_advertised_address with an avp variable but it does not replace with the avp value.... I then find out this thread https://lists.kamailio.org/pipermail/sr-users/2016-March/092327.html, in which is mentioned: *"I expect those functions not to support pvs, but I think this issomething to address in 5.0 and get all core functions working with pvs."* Looks like this was actually not done.... is that correct? Is there any plan to do it? In my case I want to use the set_advertised_address after reading the value from DB, so I can't use the advertised_address or listen options.... Any idea on how I could achieve that? Thank you, Kind regards, Patrick Wakano

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Sending INVITE from loopback subnet
by Joonas Kamailio 08 Oct '18

08 Oct '18

Dear List, I am trying to achieve a scenario where I can recognize the original peer and mask it as peer from subnet 10.0.7.0/24 with kamailio. Basic IP connectivity is established with the following network configuration (Debian9): auto lo:1 iface lo:1 inet static address 10.0.0.0/24 And of course I have routed 10.0.0.0/24 to servers public IP address (hereby known as x.x.x.1) in my router. Please consider the following scenario: - Peer1 192.168.0.1 is sending INVITE to kamailio at IP x.x.x.1 - Kamailio sends INVITE to peer2 at 172.10.0.1 from socket 10.0.0.1 - Peer2 receives the INVITE and knows that since it came from IP 10.0.0.1 it means that INVITE is originally from peer1 - IF peer2 sends INVITE to kamailio, it should be sent to peerX from socket 10.0.0.2 This works if I list both 10.0.0.1 and 10.0.0.2 with 'listen=' parameter but things get out of hand if I have multiple peers. In my tests I have noticed that 60 peers equals 60 listen parameters which equals roughly 500 kamailio processes. At some point there are just too many processes and everything melts. If I omit the listen parameter altogether, kamailio has only one socket from /24 subnet and it is 10.0.0.0. Error message when listen parameter is omitted: [pv_core.c:2612]: pv_set_force_sock(): no socket found to match [10.0.0.1] My kamailio has the following configuration in request_route: $avp(e_private) = $(sht(get_private=>$si)); $fs = $avp(e_private); I know that the case is a bit confusing and totally something normal people won't do. If you have any questions I am happy to answer. Best Regards, Joonas Keskitalo

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Server_id expects number only
by Denys Pozniak 08 Oct '18

08 Oct '18

Hello! I need to separate Kamailio's dmq_usrloc by server_id and will be much easily if server_id can accept string values also, like: server_id="192.168.100.100" But currently it accepts only integer: https://www.kamailio.org/wiki/cookbooks/5.1.x/core#server_id Syslog: 0(11488) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/global.cfg, line 126, column 11-24: syntax error 0(11488) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/global.cfg, line 126, column 11-24: number expected -- BR, Denys Pozniak

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High number of aborted mysql connections
by Sergiu Pojoga 08 Oct '18

08 Oct '18

Hi all, I'm trying to understand why some Kamailio processes become Connor MacLeod and go to live forever causing high # of aborted connections. While watching in time over *SHOW PROCESSLIST*, some 12 kamailio connections in *SLEEP* state, most of them are reused over time (time value gets reset). However, a select few go on to SLEEP forever for as much as the mysql's *wait_timeout* allows them to. Default was 28800 sec, that's how long they lived. If I lower this to say 600 sec, even higher abort rate is observed. I suppose my questions are: 1. Is this normal to have so many SLEEPing connections? 2. What is the recommended MySQL *wait_timeout* value under Kamailio? 3. Any other suggestions as to why such a high rate of aborted connections? Merci!

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Intercepting local reply attributes
by Alex Balashov 06 Oct '18

06 Oct '18

Hi, I have Kamailio in a weird interop/adaptation layer role with the following call flow: SUBSCRIBE to Kamailio with Expires: 0 ---> <-- immediate 200 OK from Kamailio [some time passes] <-- NOTIFY with additional info from Kamailio 200 OK to Kamailio ---> The NOTIFY follows a series of lengthy async HTTP interrogations, so it's constructed at a later point in time with $uac_req(...). The problem is that the standard which regulates this is extremely eccentric in the following way: despite the immediate expiration of the subscription, the NOTIFY is expected to conform to the format of an in-dialog request as if a dialog existed. Don't ask. It's as idiotic as it sounds. But it's a Broadsoft "standard". I can match the To URI / To-tag by capturing it from the incoming From header on the SUBSCRIBE and reproducing it into the outgoing To header on the NOTIFY. The issue is that my From-tag on the NOTIFY does not match the To-tag on the 200 OK for the SUBSCRIBE. So, from the point of view of the subscriber, the NOTIFY is "From" a different party to the one that "initiated" a "dialog" in response to its subscription. I'm just sending the "200 OK" for the SUBSCRIBE like this: --- append_to_reply("Event: $hdr(Event)\r\n"); append_to_reply("Expires: $hdr(Expires)\r\n"); append_to_reply("Contact: <sip:$sel(cfg_get.domain.sip_domain):$Rp>\r\n"); send_reply("200", "OK"); --- I am not aware of anything like an onsend_route that I can use to gain access to the To header of this reply. Nor am I aware of any means of removing headers from a reply and injecting my own To header with a self-generated To-tag I can later mimic in the From tag of the NOTIFY. Any creative workarounds would be appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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kamailio + rtpproxy behind NAT
by Bonjour Madame 05 Oct '18

05 Oct '18

Hello, I have kamailio and rtpproxy running on the same server located behind a router with NAT just like below. UA(internet)---public ip---router(NAT)--Kamailio+RTPproxy(private ip)--UA(private network) There are several issues like one way audio. I need users on the internet to communicate with each other, users on the private network to communicate with each other, and users on the internet and the private network to communicate with each other. What is the best approach? Best regards, Clarence

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SIP communication does not reach phone when STUN is used
by Kevin Olbrich 05 Oct '18

05 Oct '18

Hi! I noticed a minor problem when I set up the STUN config of my phone. When the server is set, I am unable to reach the endpoint from kamailio (Phone1 -> Kamailio -> Asterisk -> same Kamilio -> no outgoing SIP flow to Phone2). If I do not set it, it seems to fallback to it's RFC1918 address which is then detected by nat_uac_test() and I can reach the phone. During my tests with Asterisk directly, this did not make any difference. Seems like Asterisk checks more than FLAG 19/8 in nat_uac_test(). If I am right, I strictly used the SIP source port for answers instead of any self-detection using STUN or similar. My kamailio node is connected to the internet directly and 95 percent of (planned) phones are behind random NATs. In example kamailio.cfg there are these settings: # Caller NAT detection > route[NATDETECT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (is_method("REGISTER")) { > fix_nated_register(); > } else { > if(is_first_hop()) { > set_contact_alias(); > } > } > setflag(FLT_NATS); > } > #!endif > return; > } AND (modified for direct RTP flow): # RTPProxy control and signaling updates for NAT traversal > route[NATMANAGE] { > #!ifdef WITH_NAT > if (is_request()) { > if(has_totag()) { > if(check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > } > } > if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; > if(nat_uac_test("8")) { > fix_nated_sdp("15"); > if (is_request()) { > if (!has_totag()) { > if(t_is_branch_route()) { > add_rr_param(";nat=yes"); > } > } > } > if (is_reply()) { > if(isbflagset(FLB_NATB)) { > if(is_first_hop()) > set_contact_alias(); > } > } > #!endif > return; > } I checked the docs but I am not sure which FLAGs are safe to use (or break things). Any thoughts or hints are welcome. Thank you very much! Kind regards Kevin

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Re: [SR-Users] Kamailio as TCP UDP proxy
by M K 05 Oct '18

05 Oct '18

Thanks, that was very helpful. But now I get an "ERROR: <core> [tcp_main.c:4070]: handle_tcpconn_ev(): connect <Client IP>:<Port> failed" displayed when trying to register. The SIP client establishes the connection to the proxy via TCP, but does not receive an answer. Could this indicate a faulty NAT-ALG?

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clang compiler warnings
by Juha Heinanen 05 Oct '18

05 Oct '18

I just made a test and, for sure, clang warns about unused variables: $ clang -Wunused-variable -I/usr/include/re -I/usr/include/baresip -lbaresip -lrem -lre baresip.c baresip.c:48:6: warning: unused variable 'unused' [-Wunused-variable] int unused; ^ 1 warning generated. -- Juha

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Kamailio memory manager
by Igor Olhovskiy 05 Oct '18

05 Oct '18

Hi! What is the best way to debug messages like... 29(42) ERROR: [core/mem/q_malloc.c:290]: qm_find_free(): qm_find_free(0x7fad301cf010, 56); Free fragment not found! 29(42) ERROR: [core/mem/q_malloc.c:423]: qm_malloc(): qm_malloc(0x7fad301cf010, 56) called from core: core/data_lump.c: dup_lump_list_r(537), module: core; Free fragment not found! 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Cause core.shmmem shows total: 536870912 free: 522859104 used: 12224144 real_used: 14011808 max_used: 14479048 fragments: 10560 Also mod.stats all pkg Module: core { init_io_wait(475): 6960 init_io_wait(524): 6312 new_db_id(272): 512 dupl_string_name(72): 32 db_do_init2(298): 488 mk_switch_cond_table(43): 144 sr_wtimer_add(330): 48 as_asciiz(161): 24 init_modules(1026): 16 rpc_hash_add(146): 2048 db_allocate_columns(150): 32 db_new_result(114): 56 get_hdr_field(116): 1160 db_allocate_columns(160): 16 cnt_hash_add(383): 1024 register_select_table(458): 48 add_callback(59): 320 sr_wtimer_init(308): 136 cfg_new_group(79): 704 cfg_declare(50): 3360 fix_sock_str(420): 480 init_dst_set(83): 32208 grp_hash_add(234): 808 fix_hostname(1301): 152 add_alias(91): 464 add_alias(93): 256 dupl_string(48): 176 parse_headers(325): 432 subst_parser(291): 144 subst_parser(274): 64 exp_optimize_right(411): 24 fix_expr(541): 600 mk_case_stm(3775): 400 fixup_regexp_null(213): 72 mk_elem(90): 560 route_new_list(200): 1344 fix_param(1196): 1336 parse_select(204): 1488 fix_param(1257): 536 fix_match_rve(3028): 320 mk_rval_expr1(2616): 142656 tr_new(1646): 1464 fixup_pvar_all(279): 1624 fix_param(1162): 16144 mk_action(118): 350168 mk_rval_expr2(2674): 489552 mk_rval_expr_v(2537): 1475600 yyparse(2757): 8792 pv_parse_format(1150): 89064 pv_cache_add(347): 31456 parse_params2(585): 1400 set_mod_param_regex(110): 376 set_mod_param_regex(121): 832 tr_table_add(1875): 400 sr_cmd_exports_convert(235): 36288 register_module(280): 6624 load_module(496): 2640 new_sock_info(235): 224 fix_socket_list(1509): 48 new_sock_info(230): 3920 add_alias(93): 56 add_alias(91): 96 yyparse(1774): 64 yyparse(667): 152 yyparse(2306): 128 yyparse(648): 144 sr_push_yy_state(1666): 32 sr_push_yy_state(1656): 104 sr_push_yy_state(1596): 184 addstr(1413): 415752 rpc_hash_add(100): 25656 str_hash_alloc(59): 512 pv_table_add(236): 17864 pv_init_buffer(2055): 327680 pv_init_buffer(2045): 320 init_nonsip_hooks(43): 24 init_rlist(146): 8 mk_rval_expr_v(2548): 1264 route_add(124): 5872 str_hash_alloc(59): 768 rval_get_str(1256): 56632 pp_define(1788): 2248 init_counters(122): 128 cnt_hash_add(332): 20608 str_hash_alloc(59): 1280 Total: 3602152 } Module: rtpengine { child_init(1812): 256 Total: 256 } Module: db_mysql { db_mysql_new_connection(65): 96 db_mysql_new_connection(75): 3040 Total: 3136 } Module: siptrace { mod_init(344): 384 Total: 384 } Module: sl { sl_register_callback(460): 88 Total: 88 } Module: acc { parse_acc_extra(116): 1512 Total: 1512 } Module: dialog { dlg_bridge_init_hdrs(66): 80 Total: 80 } Module: rr { register_rrcb(61): 80 Total: 80 } Module: db_text { dbt_get_columns(67): 64 Total: 64 } Module: permissions { get_pathname(242): 64 Total: 64 } Module: auth { generate_random_secret(239): 32 generate_random_secret(238): 32 Total: 64 } Module: xlog { mod_init(214): 4104 xdbg_fixup_helper(500): 6032 xlog_fixup_helper(535): 26992 Total: 37128 } Module: dmq { make_socket_str_from_uri(149): 24 Total: 24 } Module: sqlops { sqlops_tr_buffer_init(46): 2048 sql_init_con(83): 240 Total: 2288 } Module: textops { tr_txt_parse_re(212): 32 hname_fixup(3217): 768 fixup_method(3287): 1384 Total: 2184 } Module: pv { tr_parse_string(2343): 120 tr_parse_string(2332): 32 tr_parse_string(2378): 96 tr_parse_string(2368): 184 tr_parse_string(2544): 576 add_var(58): 3616 add_var(65): 1112 Total: 5736 } Module: textopsx { fixup_hname_param(598): 352 Total: 352 } Module: avpops { avpops_parse_pvar(57): 416 Total: 416 } Module: userblacklist { check_blacklist_fixup(522): 16 Total: 16 } Module: htable { fixup_ht_key(348): 176 pv_parse_ht_name(158): 520 Total: 696 } Module: dialplan { dp_trans_fixup(396): 296 Total: 296 } Module: dispatcher { Total: 0 } Module: tmx { Total: 0 } Module: tm { Total: 0 } Module: kex { Total: 0 } Module: usrloc { Total: 0 } Module: topos { Total: 0 } Module: utils { Total: 0 } Module: app_python { Total: 0 } Module: cfgutils { Total: 0 } Module: nathelper { Total: 0 } Where is the place to look at? Means what module can cause this? Regards, Igor

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Astricon: OpenSource Soiree
by Giovanni Maruzzelli 05 Oct '18

05 Oct '18

Hello friends and collegues, A nice get together has been organized for evening Wednesday 10th at 7pm The idea is to have drinks and chitchat about opensource telephony and webrtc, and all opensource project founders, members, and practitioners are invited. Please join me, as distinguished participants too! Check it out at http://party.officering.com (no animals has been harmed in this mail) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Kamailio as TCP UDP proxy
by M K 05 Oct '18

05 Oct '18

Hi, I'm struggling with the configuration to use Kamailio as TCP UDP proxy: [SIP Client] <-(TCP)-> [Proxy] <-(UDP)-> [SIP Server] should be the setup. I only have one client and one SIP server. How do I get Kamailio configured as a proxy so that translation between TCP and UDP works (and vice versa)? I have not yet found any documentation and examples that have caused messages to arrive at my SIP server. I think I've been on the wrong track so far and I'm looking forward to any help. Thank you in advance, Laqoore

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Development for Kamailio v5.2 series is frozen
by Daniel-Constantin Mierla 05 Oct '18

05 Oct '18

Hello, short note to mark the freezing of development for Kamailio v5.2 series. For few weeks, please do not push new features in the master branch. Once the branch 5.2 is created (expected to happen in 3-4 weeks from now), the master branch becomes again open for new feature. Meanwhile the focus has to be on testing current code. Work on related tools (e.g., kamctl) or documentation can still be done as well as getting the new modules in 5.2 in good shape, plus adding exports to kemi interface (which should not interfere with old code). The entire testing phase is expected to be 4 to 6 weeks, then the release of v5.2.0 will be out. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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PATH when used with multiple listening ports
by Sergiu Pojoga 04 Oct '18

04 Oct '18

Hi again, When acting as an intermediary proxy for a non-colocated Registrar, the REGISTER request coming in on one port and going out on another, shouldn't the *add_path[received]()* generate *two* nodes in the Path header? Otherwise, how are the subsequent requests supposed to be routed back to the UAC? Scenario with Kamailio proxy on IP 1.1.1.1 with dual port listeners UAC => proxy:5070 <=> proxy:5060 => Registrar:5060 Path: <sip:1.1.1.1;lr;received=sip:123.123.123.123:59593> I would expect it to be something like: Path: <sip:1.1.1.1;lr>,<sip:1.1.1.1:5070;lr;received=sip: 123.123.123.123:59593> modparam("rr", "enable_double_rr", 1) modparam("path", "use_received", 1) Merci!

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Kamailio in a mobile world - ts_append issue
by Dan Quinney 04 Oct '18

04 Oct '18

Using "Kamailio in a mobile world" as a base, I have the following routes to store information about the current transaction, however, if the transaction is already cancelled registrations for the user will still trigger `ts_append`, raising an error as the htable hasn't expired yet. Is there anyway to remove the htable entry when the original transaction cancels? Or something else that I'm missing?

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Kamailio v5.1.6 Released
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, Kamailio SIP Server v5.1.6 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.6. For more details about version 5.1.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/10/kamailio-v5-1-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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Roadmap to next major release - v5.2.0
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, as it was discussed during our last IRC devel meeting before the summer, the time to release a new major version should be this autumn. With the Kamailio Developers Meeting planned for September 27-28, 2018, I think it is better to freeze the code couple of days after that event. So I propose the following milestones: * on Thursday, October 4, 2018 - last development day, code for 5.2.0 will be frozen at the end of the day * 4 to 6 weeks of testing * release sometime in November, to be decided more precisely after the first weeks of testing, some rough candidates being within the weeks of Nov 5 or Nov 19 If anyone has something to add or adjust, let's discuss on sr-users mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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Planing the release of Kamailio v5.1.6
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, a matter of the schedule to be agreed in the morning and the availability during the Kamailio Developers Meeting, we may do a new stable release from branch 5.1.6 on Thursday, Sep 27, 2018, in the afternoon. Given that several of us will be in the same room, the special goal would be to try to automate the process as much as possible. If the time doesn't allow it, then the release will be done next week, a specific day to be decided by the end of this week. Anyhow, this is the informal notification to report as soon as possible any issues you are aware of and not yet in bug tracker, so they get a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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tcp Free fragment not found
by Diego Nadares 04 Oct '18

04 Oct '18

Hi guys! We have four a scripts that sends dlg.list command to kamailio through xmlrpc. They are not executed simultaneously. Sometimes, we are getting the following errors: fm_search_defrag(): fm_search_defrag(0x7f212c25a000, 5273984); Free fragment not found!2017-05-26T13:54:20.438736-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f212c25a000, 5273984) called from core: tcp_main.c: tcpconn_new(957), module: core; Free fragment not found!2017-05-26T13:54:20.439101-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:959]: tcpconn_new(): mem. allocation failure2017-05-26T13:54:20.439344-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:3985]: handle_new_connect(): tcpconn_new failed, closing socket I all ready increased this two variables: tcp_conn_wq_max=30000000 tcp_rd_buf_size=5273226 How can I see how much memory xmlrpc is using? Any advice will be appreciated. Thanks in advance! Diego

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Freeing already Freed pointer
by Hamid Hashmi 04 Oct '18

04 Oct '18

I have been receiving CRITICAL core logs since I have updated my service to the following version. Print out of kamailio internals Version: kamailio 5.1.5 (x86_64/linux) d06080 Default config: /usr/local/SIPSer/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/SIPSer/lib64/kamailio/modules Compile 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 MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: d06080 Compiled with: gcc 4.8.5 Compiled on: 14:17:56 Sep 13 2018 Thank you for flying kamailio! I have been starting kamailio with the following command $ /usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamailio/kamailio.cfg -m 1024 -M 512 Sample Critical Logs are below Sep 27 14:40:19 kamailio_eu SIPSer[11289]: 923226363266 923000000001 27872 INVITE haYKxpYM1QR8XTU37WdVbxf4VkEITe7A CRITICAL: <core> [core/mem/q_malloc.c:504]: qm_free(): BUG: freeing already freed pointer (0x7f30c8a4df30), called from tm: t_reply.c: relay_reply(1920), first free tm: h_table.c: free_cell_helper(155) - aborting Sep 27 14:40:19 kamailio_eu SIPSer[11289]: 923226363266 923000000001 27872 INVITE haYKxpYM1QR8XTU37WdVbxf4VkEITe7A CRITICAL: <core> [core/mem/q_malloc.c:504]: qm_free(): BUG: freeing already freed pointer (0x7f30c8a4cc40), called from tm: h_table.c: free_cell_helper(153), first free tm: h_table.c: free_cell_helper(153) - aborting Its also causing service crash. Complete logs are here<https://paste.ee/p/q65DE>. Regards Hamid R. Hashmi

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is_first_hop() fails to detect 'myself' when using advertise
by Sergiu Pojoga 04 Oct '18

04 Oct '18

Hi ppl, If I add a FQDN advertise address to the listen statement, *is_first_hop() <https://kamailio.org/docs/modules/4.3.x/modules/siputils.html#siputils.f.is…>* seems to return FALSE no matter how I play with the listen and alias settings. listen=udp:1.1.1.1:5060 advertise sip.mydomain.net:5060 alias="sip.mydomain.net:5060" This statement doesn't get executed in the *onreply_route* *onreply_route[MANAGE_REPLY] {* *if(is_first_hop())* *set_contact_alias(); # contact alias doesn't get set?* *}* *is_first_hop()* *"The function returns true if the proxy is first hop after the original sender. For incoming SIP replies, it means that top Record-Route URI is 'myself' and source address is not matching it"* *This is the reply.* 2018/10/03 17:03:38.276846 2.2.2.2:62033 -> 1.1.1.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP sip.mydomain.net:5060 ;branch=z9hG4bK49b3.bfbe15.0;received=1.1.1.1 Via: SIP/2.0/UDP 10.10.10.10:5060;rport=5060;branch=z9hG4bK1af96136 *Record-Route*: <sip:sip.mydomain.net;lr=on;did=4cc.734> Contact: <sip:888@10.17.0.35:62033> Suggestions? Thanks.

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What is the typical network setup for kamailio?
by Kevin Olbrich 03 Oct '18

03 Oct '18

Hi! I am working successfully with Kamailio in my lab setup where Kamailio is the SBC for Asterisk. The network layout is looking like this: SIP-Phone <== PUBLIC NET ==> Kamailio (SBC) <== PRIVATE NET ==> Asterisk <== PUBLIC NET ==> Carrier Each public network is reachable from the internet and has a local firewall with IP whitelists. The internal SIP transactions are UDP-only but for external phones I would like to also listen for TCP/TLS. For this layout to work with rtpproxy (before we move on to RTPengine), we have to enable mhomed in Kamailio. We also have some routing issues with packets leaving with the wrong IP via rtpproxy (when call between carrier and external phone needs to be bridged). Most examples show that Asterisk is deployed on the same network as the external interface of Kamailio (-> Asterisk exposed to the public network). In our tests, this works much better but I have great security concerns because this Asterisk instance itself does not need to be reachable from external. How do other users deploy Kamailio in front of Asterisk or similar as SBC to secure internals? There is lot of docs for Kamailio's config but IMHO less for the setup as DMZ (SBC) proxy. Thank you very much. Kind regards Kevin

7 15

Using TLS on a load-balancing cluster behind DNS-SRV
by Kevin Olbrich 03 Oct '18

03 Oct '18

Hi! Which hostname do I need to request for the certificate when the servers are load-balanced using DNS-SRV? Do I need to get the cert for the DNS-SRV subdomain (without _sip._tls) or for the servers, eg. server0{1,2,3}.pbx.example.com ? Thank you! Kevin

3 2

rr module not detecting nat=ws route param
by Jon Bonilla (Manwe) 02 Oct '18

02 Oct '18

Hi all. I'm having problems with an indialog ACK request which is not being routed correctly. I call loose_route and print: $du=sip:MYIP:2443;transport=ws;r2=on;lr=on;did=1d1.7702;nat=ws $route_uri=sip:MYIP;r2=on;lr=on;did=1d1.7702;nat=ws But when I call check_route_param("nat=ws") I get false result. version: kamailio 5.1.4 any hints? cheers, Jon -- https://pekepbx.com https://www.issabel.com/multitenant

3 6

Dialog lifetime for dialogs synced with DMQ
by jenus＠cyberchaos.nl 02 Oct '18

02 Oct '18

Hello, I'm using DMQ to sync dialogs between a active and standby kamailio node. I noticed that when the standby takes over dialog's they do not expire after the lifetime i set on the other node. Is there a way to make sure that calls are terminated after the dialog lifetime expires? I see that the timer is replicated to the other node, but nothing happens after the dialog expires. The event routes aren't executed as well. Thanks, Jan

2 1

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